Alcatel-Lucent OMNIACCES-VOIP GATEWAY User Manual

VoIP User Manual

Release 4.5

An Alcatel service agreement brings your company the assurance of 7x24 no-excuses technical support. You’ll also receive regular software updates to maintain and maximize your Alcatel product’s features and functionality and on-site hardware replacement through our global network of highly qualified service delivery partners. Additionally, with 24-hour-a-day access to Alcatel’s Service and Support web page, you’ll be able to view and update any case (open or closed) that you have reported to Alcatel’s technical support, open a new case or access helpful release notes, technical bulletins, and manuals. For more information on Alcatel’s Service Programs, see our web page at www.ind.alcatel.com, call us at 1-800-995-2696, or email us at support@ind.alcatel.com.

This manual documents Release 4.5 Voice over IP (VoIP) hardware and software.

The functionality described in this manual is subject to change without notice.

Alcatel® and the Alcatel logo are registered trademarks of Alcatel. Xylan®, OmniSwitch®, PizzaSwitch® and OmniStack® are registered trademarks of Alcatel Internetworking, Inc.

AutoTracker™, OmniAccess™, OmniCore™, Omni Switch/Router™, OmniVista™, PizzaPort™, PolicyView™, RouterView™, SwitchManager™, SwitchStart™, VoiceView™, WANView™, WebView™, X-Cell™, X-Vision™ and the Xylan logo are trademarks of Alcatel Internetworking, Inc.

All-In-One of their respective companies.

is a service mark of Alcatel Internetworking, Inc. All other brand and product names are trademarks

26801 West Agoura Road

Calabasas, CA 91301

(818) 880-3500 FAX (818) 880-3505

info@ind.alcatel.com

US Customer Support–(800) 995-2696

International Customer Support–(818) 878-4507

Internet–http://eservice.ind.alcatel.com

Cautions

FCC Compliance

digital device pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions in this guide, may cause interference to radio communications. Operation of this equipment in a residential area is likely to cause interference, in which case the user will be required to correct the interference at his own expense.

The user is cautioned that changes and modifications made to the equipment without approval of the manufacturer could void the user’s authority to operate this equipment. It is suggested that the user use only shielded and grounded cables to ensure compliance with FCC Rules.

This equipment does not exceed Class A limits per radio noise emissions for digital apparatus, set out in the Radio Interference Regulation of the Canadian Department of Communications.

Avis de conformité aux normes du ministére des Communications du Canada

Cet équipement ne dépasse pas les limites de Classe A d’émission de bruits radioélectriques pour les appareils numériques, telles que prescrites par le Réglement sur le brouillage radioélectrique établi par le ministére des Communications du Canada.

Lithium Batteries Caution

incorrectly replaced. Replace the battery only with the same or equivalent type of battery recommended by the manufacturer. Dispose of used batteries according to the manufacturer’s instructions. The manufacturer’s instructions are as follows:

Return the module with the Lithium battery to Alcatel. The Lithium battery will be replaced at Alcatel’s factory.

: This equipment has been tested and found to comply with the limits for Class A

: There is a danger of explosion if the Lithium battery in your chassis is

page iii

page iv

1 VoIP Overview

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1

VoIP Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1

Getting Started with VoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-2

VoIP Telephone Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-4

A VoIP Call Scenario . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-4

Elements of a Converged Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-7

VoIP H.323 Client . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-7

VoIP H.323 Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-7

VoIP H.323 Gatekeeper (Optional) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-8

H.323 VoIP Gateway Voice and Convergence Features . . . . . . . . . . . . . . . . . . . . 1-10

Signaling Control and Voice Interoperability (Voice Features) . . . . . . . . . . . . 1-11

H.323 Call Control and Network Interoperability (Convergence Features) . . . 1-15

H.323 Network Call Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-15

Alcatel VoIP Network Dialing Schemes (AVNDS) . . . . . . . . . . . . . . . . . . 1-15

Switch Backplane Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-16

VoIP Standards for Development . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-17

Codec Support (G.711, G.723.1, G.729a) . . . . . . . . . . . . . . . . . . . . . . . . . 1-17

VON (Voice on the Net) Developments . . . . . . . . . . . . . . . . . . . . . . . . . 1-18

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1

VoIP and VLANs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-19

2 VoIP Daughtercards

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1

VoIP Daughtercard Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-2

Digital Signal Processors (DSPs), DIMMs and Available Channels . . . . . . . . 2-4

Voice Switching Daughtercard — Digital . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-6

VSD Front Panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-7

VSD Deadman Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-8

VSD Cross-Over Toggle Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-9

VSD Pinouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-11

VSD Jumpers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-12

Voice Switching Daughtercard — Euro BRI ISDN . . . . . . . . . . . . . . . . . . . . . . . . 2-13

Digital Signal Processors (DSPs) and Available Channels . . . . . . . . . . . . . 2-13

VSB Deadman Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-13

VSB NT (LT)/TE Cross-Over Toggle Switch . . . . . . . . . . . . . . . . . . . . . . . 2-13

VSB Pinouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-13

VSB Front Panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-15

VSB Jumpers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-16

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1

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Table of Contents

Voice Switching Daughtercard — Analog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-19

VSA Front Panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-21

VSA Pinouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-22

VSAs and Digital Signal Processors (DSPs), DIMMs and Available

Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-22

VSAs and the Deadman Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-22

VSAs and Cross-Over Toggle Switches . . . . . . . . . . . . . . . . . . . . . . . . . . 2-22

VSA Jumpers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-23

VSX Switching Module . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-25

VSX Technical Specifications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-25

VoIP Daughtercard Port Numbering Schemes . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-28

3 Network Dialing Schemes

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-1

How to Select a Network Dialing Scheme (AVNDS) . . . . . . . . . . . . . . . . . . . . 3-3

Network Dialing Scheme VoIP Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-5

VoIP Networks without PSTN — Example 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-7

Four Digit Extensions and Two Voice Daughtercards . . . . . . . . . . . . . . . . . . . 3-7

VoIP Networks without PSTN — Example 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-9

Trunk Groups and Three Voice Daughtercards . . . . . . . . . . . . . . . . . . . . . . . . 3-9

VoIP Networks without PSTN — Example 3 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-11

One Hunt Group (48 Channels Across Two T1s) . . . . . . . . . . . . . . . . . . . . . 3-11

VoIP Networks without PSTN — Example 4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-13

One Hunt Group (60 Channels Across Two E1s) . . . . . . . . . . . . . . . . . . . . . 3-13

VoIP Networks without PSTN — Example 5 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-15

One Hunt Group (96 Channels Across Four T1s) . . . . . . . . . . . . . . . . . . . . . 3-15

VoIP Networks without PSTN — Example 6 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-17

One Hunt Group (144 Channels Across Six T1s) . . . . . . . . . . . . . . . . . . . . . . 3-17

VoIP Networks without PSTN — Example 7 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-19

Four Hunt Groups (12 Channels Per Hunt Group) . . . . . . . . . . . . . . . . . . . . 3-19

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-1

page vi

VoIP Networks without PSTN — Example 8 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-21

48 Individual Hunt Groups (One Channel Per Group) . . . . . . . . . . . . . . . . . 3-21

VoIP Networks without PSTN — Example 9 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-23

Trunk Groups and Mixed Length Extensions . . . . . . . . . . . . . . . . . . . . . . . . 3-23

VoIP Networks without PSTN — Example 10 . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-25

Strip Digit Length (2) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-25

VoIP Networks without PSTN — Example 11 . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-27

Trunk Groups and Eleven Digit Extensions . . . . . . . . . . . . . . . . . . . . . . . . . 3-27

VoIP Networks without PSTN — Example 12 . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-29

H.323 Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-29

VoIP Networks with PSTN — Example 13 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-31

North American PSTN and VoIP Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-31

Table of Contents

VoIP Networks with PSTN — Example 14 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-37

North American PSTN, International PSTN and VoIP Calls . . . . . . . . . . . . . . . 3-37

VoIP Networks with PSTN — Example 15 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-43

PSTN and Eleven Digit Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-43

VoIP Networks with PSTN — Example 16 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-45

FAX over IP Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-45

VoIP Networks with PSTN — Example 17 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-47

Mixed Digital and Analog Voice Daughtercards . . . . . . . . . . . . . . . . . . . . . . 3-47

VoIP Networks with PSTN — Example 18 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-49

Caller ID (Static) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-49

VoIP Networks with Interoperability — Example 19 . . . . . . . . . . . . . . . . . . . . . . 3-51

H.323 Gateway to Microsoft NetMeeting (without FastStart) . . . . . . . . . . . . . . 3-51

VoIP Networks with Interoperability — Example 20 . . . . . . . . . . . . . . . . . . . . . . 3-53

H.323 Gateway to Cisco Router . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-53

VoIP Networks with Interoperability — Example 21 . . . . . . . . . . . . . . . . . . . . . . 3-55

H.323 Gateway to OmniPCX 4400 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-55

VoIP Networks with Interoperability — Example 22 . . . . . . . . . . . . . . . . . . . . . . 3-57

OmniPCX 4400 and E1 QSIG . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-57

VoIP Networks with Interoperability — Example 23 . . . . . . . . . . . . . . . . . . . . . . 3-59

OmniPCX and Euro PRI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-59

VoIP Networks with Interoperability — Example 24 . . . . . . . . . . . . . . . . . . . . . . 3-61

Other PBXs with T1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-61

VoIP Networks with Interoperability — Example 25 . . . . . . . . . . . . . . . . . . . . . . 3-63

Other PBXs with Euro BRI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-63

VoIP Networks with Interoperability — Example 26 . . . . . . . . . . . . . . . . . . . . . . 3-65

Mixed European Digital and Analog Voice Daughtercards . . . . . . . . . . . . . . . 3-65

AVNDS Master List of Features by CLI Command . . . . . . . . . . . . . . . . . . . . . . . . 3-67

4 Setup and Installation

Components of VoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-1

Assumptions and Recommendations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-2

Configuration Restrictions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-3

General Installation Procedures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-4

Instructions for Additional VoIP Installations . . . . . . . . . . . . . . . . . . . . . . . . . . 4-7

Example VSM Boot File (

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-1

vsmboot.asc

) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-8

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Table of Contents

5 VoIP Commands

Voice Switching Daughtercard Commands . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-14

voice daughter card ip mask . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-16

voice daughter card ip address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-17

voice daughter card ip default gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-18

voice daughter card activate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-19

voice dump . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-20

voice daughter card h.323 out fast start . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-23

voice daughter card h.323 in fast start . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-24

voice daughter card h.323 auto answer . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-25

voice daughter card first digit wait duration . . . . . . . . . . . . . . . . . . . . . . . . . 5-26

voice daughter card inter digit wait duration . . . . . . . . . . . . . . . . . . . . . . . . . 5-27

voice daughter card dial time wait duration . . . . . . . . . . . . . . . . . . . . . . . . . 5-28

voice daughter card termination digit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-29

voice daughter card cadence coefficient . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-30

voice daughter card ring id . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-31

voice daughter card vsb clock source . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-32

voice daughter card vsb external clock port . . . . . . . . . . . . . . . . . . . . . . . . . 5-33

voice port interface type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-34

voice port frame format . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-36

voice port circuit identifier . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-38

voice port nfas framing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-39

voice port line build out . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-40

voice port line length . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-41

voice port attenuation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-42

voice port cable type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-43

voice port line coding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-44

voice port facilities data link protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-45

voice port facilities data link port role . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-46

voice port transmit clock source . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-47

voice port loop back mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-48

voice port signaling mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-49

voice port trap generation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-50

voice port isdn protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-51

voice port isdn switch type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-53

voice port bri line type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-54

voice channel isdn d channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-55

voice channel isdn b channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-57

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-1

page viii

Channel Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-58

voice channel mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-59

voice channel dial in private line automatic ringdown . . . . . . . . . . . . . . . . . . 5-60

voice channel state . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-61

Table of Contents

Telephony Signaling Attributes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-62

view voice signaling channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-67

voice signaling protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-68

voice signaling out wait . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-69

voice signaling out tone digit duration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-70

voice signaling out tone interdigit duration . . . . . . . . . . . . . . . . . . . . . . . . . . 5-71

voice signaling out dialing type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-72

voice signaling call duration limit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-73

voice signaling answer wait limit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-74

voice signaling hang up wait limit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-75

voice signaling fax holdover . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-76

voice signaling companding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-77

voice signaling receive gain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-78

voice signaling transmit gain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-79

voice signaling idle noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-80

voice signaling em on hook debounce . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-81

voice signaling em off hook debounce . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-82

voice signaling em seize detect . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-83

voice signaling em clear detect . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-84

voice signaling em clear confirm detect . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-85

voice signaling em clear confirm wait max . . . . . . . . . . . . . . . . . . . . . . . . . . 5-86

voice signaling em guard all . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-87

voice signaling em guard out . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-88

voice signaling em dial tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-89

voice signaling em min connection time . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-90

voice signaling em hang up wait . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-91

voice signaling emw in wink wait min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-92

voice signaling emw in wink wait max . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-93

voice signaling emw in wink duration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-94

voice signaling emw in wink digit ignore . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-95

voice signaling emw out wink wait max . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-96

voice signaling emw out wink duration min . . . . . . . . . . . . . . . . . . . . . . . . . 5-97

voice signaling emw out wink duration max . . . . . . . . . . . . . . . . . . . . . . . . 5-98

voice signaling emi glare report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-99

voice signaling emi digit wait . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-100

voice signaling emd in delay min . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-101

voice signaling emd in delay max . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-102

voice signaling emd in digit ignore . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-103

voice signaling emd out integrity check . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-104

voice signaling emd out delay duration min . . . . . . . . . . . . . . . . . . . . . . . . . 5-105

voice signaling emd out detail duration max . . . . . . . . . . . . . . . . . . . . . . . . .5-106

voice signaling emd out delay check . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-107

voice signaling fxs ls on hook debounce . . . . . . . . . . . . . . . . . . . . . . . . . . .5-108

voice signaling fxs ls off hook debounce . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-109

voice signaling fxs ls seize detect . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-110

voice signaling fxs ls originate clear detect . . . . . . . . . . . . . . . . . . . . . . . . . .5-111

voice signaling fxs ls answer clear detect . . . . . . . . . . . . . . . . . . . . . . . . . . .5-112

voice signaling fxs ls supervisory disconnect wait . . . . . . . . . . . . . . . . . . . . .5-113

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voice signaling fxs ls supervisory disconnect duration . . . . . . . . . . . . . . . . . .5-114

voice signaling fxs ls caller id . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-115

voice signaling fxs ls ringing debounce . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-116

voice signaling fxo ls supervisory disconnect detection . . . . . . . . . . . . . . . . .5-117

voice signaling fxo ls supervisory disconnect . . . . . . . . . . . . . . . . . . . . . . . . 5-118

voice signaling fxo ls guard out . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-119

voice signaling fxo ls ringing inter cycle . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-120

voice signaling fxo ls ringing inter pulse . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-121

voice signaling fxo ls caller id . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-122

voice signaling fxo ls answer after . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-123

voice signaling fxo ls loop current debounce . . . . . . . . . . . . . . . . . . . . . . . .5-124

voice signaling fxo ls battery reversal debounce . . . . . . . . . . . . . . . . . . . . . . 5-125

voice signaling fxs gs seize detect . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-126

voice signaling fxs gs on hook debounce . . . . . . . . . . . . . . . . . . . . . . . . . . .5-127

voice signaling fxs gs originate clear detect . . . . . . . . . . . . . . . . . . . . . . . . .5-128

voice signaling fxs gs answer clear detect . . . . . . . . . . . . . . . . . . . . . . . . . . .5-129

voice signaling fxs gs min ring ground . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-130

voice signaling fxs gs max wait loop . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-131

voice signaling fxs gs min loop open . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-132

voice signaling fxs gs caller id . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-133

voice signaling fxs gs off hook debounce . . . . . . . . . . . . . . . . . . . . . . . . . . .5-134

voice signaling fxs gs ring ground debounce . . . . . . . . . . . . . . . . . . . . . . . .5-135

voice signaling fxs gs ring id . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-136

voice signaling fxo gs connection loop open debounce . . . . . . . . . . . . . . . .5-137

voice signaling fxo gs max tip ground wait . . . . . . . . . . . . . . . . . . . . . . . . . .5-138

voice signaling fxo gs tip ground debounce . . . . . . . . . . . . . . . . . . . . . . . . .5-139

voice signaling fxo gs ringing debounce . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-140

voice signaling fxo gs ringing inter cycle . . . . . . . . . . . . . . . . . . . . . . . . . . .5-141

voice signaling fxo gs ringing inter pulse . . . . . . . . . . . . . . . . . . . . . . . . . . .5-142

voice signaling fxo gs caller id detection . . . . . . . . . . . . . . . . . . . . . . . . . . .5-143

voice signaling fxo gs answer after . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-144

voice signaling fxo gs loop current debounce . . . . . . . . . . . . . . . . . . . . . . . .5-145

voice signaling fxo gs battery reversal debounce . . . . . . . . . . . . . . . . . . . . . 5-146

voice signaling caller id name . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-147

voice signaling caller id number . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-148

voice signaling tone table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-149

voice signaling call progress tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-150

voice signaling call progress tone detection configuration . . . . . . . . . . . . . . .5-151

voice signaling v.18 tone detection threshold hang time . . . . . . . . . . . . . . . .5-152

voice signaling v.18 tone detection threshold level . . . . . . . . . . . . . . . . . . . .5-153

voice signaling v.18 tone detection threshold fraction . . . . . . . . . . . . . . . . . . 5-154

voice signaling single frequency tone detection threshold level . . . . . . . . . . .5-155

voice signaling single frequency tone detection threshold time . . . . . . . . . . .5-156

voice signaling echo canceller non-linear sensitivity . . . . . . . . . . . . . . . . . . .5-157

voice signaling acoustic echo canceller mode . . . . . . . . . . . . . . . . . . . . . . . .5-158

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voice signaling acoustic echo canceller non-linear processor . . . . . . . . . . . . .5-159

voice signaling acoustic echo canceller output . . . . . . . . . . . . . . . . . . . . . . .5-160

voice signaling acoustic echo canceller handset speaker gain . . . . . . . . . . . .5-161

voice signaling acoustic echo canceller hands free speaker gain . . . . . . . . . .5-162

voice signaling override in band call progress tones . . . . . . . . . . . . . . . . . . .5-163

voice signaling override full call progress tones . . . . . . . . . . . . . . . . . . . . . .5-164

voice signaling override ring back . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-165

voice signaling override in band codec switching . . . . . . . . . . . . . . . . . . . . .5-166

voice signaling override psu codec switching . . . . . . . . . . . . . . . . . . . . . . . . 5-167

voice signaling override network overlap dialing . . . . . . . . . . . . . . . . . . . . .5-168

voice signaling override information element transport . . . . . . . . . . . . . . . . . 5-169

voice signaling override qsig information element transport . . . . . . . . . . . . .5-170

voice signaling override setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-171

Coding Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-172

voice coding profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-174

voice no coding profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-175

view voice coding profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-176

voice coding profile all reset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-180

voice channel available coding profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-181

voice channel assign preferred coding profile . . . . . . . . . . . . . . . . . . . . . . . .5-182

voice coding profile coding type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-183

voice coding profile voice packet interval . . . . . . . . . . . . . . . . . . . . . . . . . . .5-185

voice coding profile voice network delay buffer mode . . . . . . . . . . . . . . . . .5-186

voice coding profile voice network delay buffer nominal delay . . . . . . . . . . .5-187

voice coding profile voice network delay buffer max delay . . . . . . . . . . . . . .5-188

voice coding profile voice activity detector . . . . . . . . . . . . . . . . . . . . . . . . . .5-189

voice coding profile voice activity detection threshold mode . . . . . . . . . . . . .5-190

voice coding profile voice activity detection threshold level . . . . . . . . . . . . .5-191

voice coding profile voice dtmf relay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-192

voice coding profile switchover . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-193

voice coding profile call progress tone detection . . . . . . . . . . . . . . . . . . . . .5-194

voice coding profile voice dtmf relay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-195

voice coding profile single frequency tone detection . . . . . . . . . . . . . . . . . .5-196

voice coding profile voice echo canceller . . . . . . . . . . . . . . . . . . . . . . . . . . .5-197

voice coding profile voice echo canceller non linear . . . . . . . . . . . . . . . . . . .5-198

voice coding profile voice echo canceller comfort noise mode . . . . . . . . . . .5-199

voice coding profile echo canceller noise level . . . . . . . . . . . . . . . . . . . . . . .5-200

voice coding profile voice echo canceller tail length . . . . . . . . . . . . . . . . . . .5-201

voice coding profile echo canceller refresh configuration . . . . . . . . . . . . . . .5-202

voice coding profile echo canceller refresh state . . . . . . . . . . . . . . . . . . . . . .5-203

voice coding profile fax rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-204

voice coding profile fax transmit level . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-205

voice coding profile fax carrier detect threshold . . . . . . . . . . . . . . . . . . . . . .5-206

voice coding profile fax timeout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-207

voice coding profile fax t.38 high speed packet rate . . . . . . . . . . . . . . . . . . .5-208

voice coding profile fax t.38 low speed redundancy . . . . . . . . . . . . . . . . . . .5-209

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voice coding profile fax t.38 high speed redundancy . . . . . . . . . . . . . . . . . .5-210

voice coding profile fax t.38 training check field method . . . . . . . . . . . . . . .5-211

voice coding profile silence detect time . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-212

voice coding profile silence detect level . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-213

voice coding profile g.711 modem resampling mode . . . . . . . . . . . . . . . . . .5-214

voice coding profile caller id . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-215

Voice Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-216

voice network card h.323 gatekeeper control . . . . . . . . . . . . . . . . . . . . . . . .5-217

voice network h.323 gatekeeper mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-218

voice network h.323 gatekeeper address . . . . . . . . . . . . . . . . . . . . . . . . . . .5-219

voice network h.323 allow calls without gatekeeper . . . . . . . . . . . . . . . . . . .5-220

voice network h.323 allow calls without gatekeeper max tries . . . . . . . . . . . .5-221

voice network h.323 endpoint registration type . . . . . . . . . . . . . . . . . . . . . . 5-222

voice network h.323 gatekeeper associate . . . . . . . . . . . . . . . . . . . . . . . . . .5-223

voice network h.323 display name . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-224

voice network h.323 rtp port mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-225

voice network h.323 rtp port base . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-226

Network Dialing Scheme . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-228

voice destination h.323 endpoint . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-230

voice destination local channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-231

voice no destination . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-232

view voice destination . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-233

voice phone group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-235

voice no phone group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-236

view voice phone group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-237

voice phone group site prefix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-238

voice phone group site prefix digits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-239

voice phone group type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-240

voice phone group format . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-243

voice phone group strip digit length . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-244

voice phone group forwarding prefix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-245

voice phone group forwarding prefix digits . . . . . . . . . . . . . . . . . . . . . . . . .5-246

voice phone group add numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-247

voice phone group delete numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-248

voice numbering plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-249

voice no numbering plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-250

view voice numbering plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-251

voice numbering plan activate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-253

voice numbering plan hunt method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-255

voice numbering plan description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-256

voice numbering plan destination member . . . . . . . . . . . . . . . . . . . . . . . . . .5-257

voice numbering plan phone group member . . . . . . . . . . . . . . . . . . . . . . . .5-258

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System-Wide VoIP Commands . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-259

view voice daughter card . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-260

view voice port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-261

view voice channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-262

view voice network card . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-263

voice daughter card statistics collection . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-264

voice channel reset all statistics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-265

view voice channel telephony level stats . . . . . . . . . . . . . . . . . . . . . . . . . . .5-266

view voice channel telephony channel stats . . . . . . . . . . . . . . . . . . . . . . . . .5-267

view voice channel voice playout stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-268

view voice channel dsp stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-269

view voice channel error stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-270

view voice channel modem stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-271

view voice channel fax stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-272

view voice channel isdn level 2 stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-273

voice channel reset telephony level stats . . . . . . . . . . . . . . . . . . . . . . . . . . .5-274

voice channel reset telephony channel stats . . . . . . . . . . . . . . . . . . . . . . . . .5-275

voice channel reset voice playout stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-276

voice channel reset dsp stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-277

voice channel reset error stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-278

voice channel reset modem stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-279

voice channel reset fax stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-280

reset voice channel isdn level 2 stats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5-281

Index

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .I-1

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Table of Contents

page xiv

1 VoIP Overview

Introduction

This chapter describes Alcatel’s H.323 Voice over IP (VoIP) gateway and how telephone or fax calls can be programmed to automatically go through either an enterprise’s Virtual Private Network (VPN) via the gateway, and/or the Public Switched Telephone Network (PSTN), a circuit-switched public telephone network that consists of all the interconnected calling networks in the world.

Alcatel’s H.323 VoIP gateway provides transparent, seamless delivery and connection of local and long distance, inbound and outbound telephone calls initiated through standard PSTN North American (T1), European (E1), and Euro ISDN (Integrated Services Digital Network) digital telephone transmission lines. For specific details on the precise types of calls handled, see Chapter 2, “VoIP Daughtercards” and Chapter 5, “VoIP Commands.”

As with standard T1, E1, ISDN (Euro) telephone service, VoIP calls can be transmitted fullduplex (simultaneously in both directions). Likewise, Alcatel’s H.323 VoIP gateway digitizes phone or fax call signals and uses one of these call transmission services, depending on the type of call, to channel the calls, i.e., carry information to a destination point in the VoIP network. Depending on the configuration of the VoIP network, the calls may also go through the PSTN. For more details on the digitizing process, see

Signal Processing

This chapter provides general background information on VoIP networks, clients, gateways and gatekeepers, and includes a list of key features provided specifically by Alcatel’s H.323 VoIP network. A VoIP call scenario is illustrated and described along with details on the technologies used in VoIP to explain how calls can be placed over IP. Elements of converged VoIP network are also shown and described, and significant telephone and data communications technologies are explained in relation to the VoIP gateway. Technical standards for the most prominent technologies used in Voice over IP are briefly summarized at the end of this chapter, since as a whole, H.323 ITU standards define the major components using VoIP technologies in network-based VoIP communications systems.

on page 1-12 for more details.

Payload Packetization and Digital

VoIP Networks

Alcatel’s H.323 VoIP gateway for packet-switched IP networks combines the speed, versatility and low cost of IP telephony with standard telephone features for enterprises in North American and Europe (this necessarily entails other continents or countries, such as Mexico, that may have the same requirements). These networks are referred to as VoIP networks. Because data networks as such usually operate with extra carrying capacity (bandwidth), most IP networks are easily able to accommodate voice/fax traffic once the VoIP network is set up.

The Internet Protocol (IP) is used mainly because it is supported over many layer 1 and 2 network technologies including Ethernet (10, 100, 1000 Mbps), Token Ring, FDDI and Frame Relay to name a few, including leased lines and satellites. Nearly every router, frame relay device, and network switch used today supports the Internet Protocol. IP delivers any transport media used between local and wide area networks.

Enterprise IP networks consist of local area networks (LANs) installed at corporate offices often joined together by corporate wide area networks (WANs). Usually the local area networks support IP on various types of standard data communications technologies such as Ethernet, Token Ring, ATM (Asynchronous Transfer Mode) and FDDI (Fiber Distributed Data Interface).

Page 1-1

Introduction

Getting Started with VoIP

Wide area networks are used to support IP connections over leased lines, public frame relay, ATM, satellite and ISDN. At each branch office location, enterprises use routers to connect the remote LANs to the IP WAN. When used with Alcatel’s H.323 VoIP gateway, these Virtual Private Networks, or VPNs, allow a portion of the Public Switched Telephone Network to be managed and used by the enterprise.

Alcatel’s H.323 VoIP gateway provides the Voice over IP network capabilities by means of either digital or analog VoIP daughtercards installed in the switch. The VoIP daughtercards specifically enable enterprises to control the call routing capabilities of their own organizations by using a portion of the PSTN.

Once an enterprise network is ready to provide VoIP using Alcatel’s H.323 VoIP gateways, Network Administrators can begin setting up VoIP networks by installing and configuring the appropriate voice switching daughtercard(s). As a whole, Alcatel’s VoIP H.323 gateways can be scaled from a minimum of two voice channels per switch to a maximum of 120 voice channels per switch. Switches with the greatest VoIP scalability will use voice switching modules (VSXs) in Omni Switch/Routers. See Chapter 2, “VoIP Daughtercards,” for further details on supported configurations and scalability.

Initially, an Alcatel VoIP network dialing scheme (AVNDS) must also be selected and deployed via a text-based configuration boot file, wherein each daughtercard must be assigned a unique IP address among other unique gateway identifiers. From that point, operational parameters such as channel and port types can be set using the command line interface (CLI) configuration tool. Comparable text-based (ASCII) configuration boot files may also be quickly generated to configure multiple VoIP-enabled switches with similar requirements.

Also, stored in the

vsmboot.asc

files are voice coding parameters which are pre-configured and kept in profiles. Coding Profiles are configured directly to the components, and define which operational VoIP characteristics will be used, and then implemented according to the instructions contained in the profiles. Coding Profiles consist of general caller information, voice and fax transmission, coding/decoding settings. Preferred Coding Profiles can be automatically selected based upon payload requirements. Coding Profiles are configured at the channel level. VoIP configurations for VoIP callers are established by setting up profiles and then assigning the profiles to each individual H.323 VoIP gateway or daughtercard. Profiles can be created, modified, copied and deleted using one of the available configuration tools. It should be known that in most circumstances, the default settings for the Coding Profiles are sufficient.

Additional parameters that require configuration include calling Destinations and Network Numbering Schemes, the latter being comprised primarily of Numbering Plans, Phone Groups and Hunt Methods. Altogether, use of these parameters enable VoIP networks to translate IP addresses from telephone numbers, and allow communications between the VoIP branch offices to be configurable. For more details, see Alcatel VoIP Network Dialing Schemes

(AVNDS) on page 1-15.

See Chapter 5, “VoIP Commands,” for details on using these CLI commands once the H.323 VoIP gateway is configured; refer to this chapter as well if using an optional third-party gatekeeper (server) i.e., NT100 RADVision, on a PC for example, as some additional network parameters will need to be set.

For details on configuring the AVNDS, see “Chapter 3, “Network Dialing Schemes” and Chapter 5, “VoIP Commands.” For details on installing the cards and setting up VoIP H.323 Gateways, see Chapter 4, “Setup and Installation.”

Page 1-2

♦

Introduction

Alcatel’s H.323 VoIP Gateway Key Features

Alcatel’s H.323 VoIP gateways, which connect voice and data networks, minimize call complexity and dependency on leased telephone lines by allowing enterprises more control over their own call processing. Alcatel’s H.323 VoIP gateway is used to transport digitized voice conversations over IP local area networks, which are then sent over wide are networks using such protocols as Frame Relay or ATM. All VoIP daughtercards are compatible with the Alcatel OmniAccess 512 and Omni Switch/Router. As shown below, the following features of Alcatel’s H.323 VoIP gateway are supported in this release.

• Digital T1/E1 voice and fax transport over IP networks

• T1 and E1 telephony interface links to digital Private Branch Exchanges (PBXs) via digital

or analog VoIP daughtercards.

• T1 robbed-bit Channel Associated Signaling (CAS)

• E1 Primary Rate Interface (Euro PRI) and E1 QSIG ISDN Common Channel Signaling (CCS)

• E1 Basic Rate Interface (Euro BRI) ISDN; (E1 ETSI)

• Foreign Exchange Station (FXS) telephony Loop interface via analog VoIP daughtercard

with (FXS) grand-daughtercard (variations includes FX Office— FXO).

• H.323 Network Call Control Gateway (establishes Local Area Network (LAN) terminal links;

performs call setup and voice translation functions; provides communications procedures between LANs)

• Voice Codecs: Pulse Code Modulation (G.711), Internet Speech (G.723.1), Standard Tele-

phone Quality (G.729A), Realtime Fax over IP (Fax T.38).

• Non-Voice Signal Monitoring, Detection and Transmission Protocols:

• Dual Tone Multi Frequency/Modem Fax Relay

• Fax Transparency and Fax Relay

• Modem Transparency and Modem Relay

• PSTN Fallback via Deadman Relay Switch

• The H.323 VoIP gateway is capable of providing PSTN fallback for VoIP calls in the event of a power failure in the VoIP network by means of a Deadman relay switch on the digital VoIP daughtercards. For more information on the Deadman switch, see Chapter 2, VoIP Daughtercards.”

• Echo and jitter controls on digital VoIP daughtercards.

• Pre-configured, modifiable AVNDS (Alcatel VoIP Network Dialing Schemes) with corresponding text-based (ASCII) configuration boot files (

vsmboot.asc

files).

• VoIP Text-based Command Line Interface (CLI) configuration tool.

Note ♦

When used separately, the terms E1 and ETSI both entail European PRI and BRI interfaces. E1 ETSI used together as one term refers specifically to Euro PRI.

Page 1-3

VoIP Telephone Calls

H.323 VoIP telephone calls, which can carry either voice, facsimile, or modem transmissions over IP networks, are switched to the packet-based network and connected to the calling destination (an IP device) via a unique IP address and local/remote dialing plan (actually two Alcatel VoIP Network Dialing Schemes rolled into one). The numerical IP address, also serving to identify calls intended for VoIP networks, is determined and translated from a destination telephone number in a phone directory database while it is being entered, and the call is in progress. (It should be noted here that callers do not need to remember the IP addresses, only the called party or destination phone number). See Chapter 3, “Network Dialing Schemes,” for more information on the AVNDS.

H.323 VoIP telephone calls are transparent so callers don’t have to worry about any special procedures, except being aware of a dialing plan that may require them to dial a prefix, such as 7, before a call can be placed across the VoIP network. This would be similar to current dialing plans requiring callers to dial 9 before an office call can be placed (9 is the prefix most often used by PBXs to access the PSTN).

VoIP calls initiated from standard telephone handsets after a preset number of digits are dialed, for example, can be immediately transmitted using IP data networks whereby digital or analog signals, meant to set up connections for carrying information, are intercepted by Alcatel’s H.323 VoIP gateways in the network. These gateways translate the phone numbers into IP addresses, convert the information to digital packet form, and then deliver the calls over the network and the PSTN as shown below.

A VoIP Call Scenario

Once a VoIP network is set up a typical VoIP call scenario might go something like this.

Local Telephone Number

Remote Telephone Number

VPN

Step

Call setup begins

Source

IP Address

Dial Tone

VoIP H.323 Gateway

PBX #1

VoIP H.323 Gateway

PBX #2

Destination

IP Address

Page 1-4

Dial Destination Number

VoIP Call Scenario — Step 1: Call Setup

VoIP Telephone Calls

Local Telephone Number

Step

Session setup with remote gateway

Source

IP Address

VoIP H.323 Gateway

VPN

Dialed digits translated to

IP address

VoIP H.323 Gateway

PBX #1

VoIP Call Scenario — Step 2: Call Progress

Remote Telephone Number

Destination

IP Address

PBX #2

As the caller dials, the H.323 VoIP gateway collects the dialed digits and then ultimately translates the digits using a pre-configured Numbering Plan and Phone Group into the IP address. A VoIP session is then initiated with the remote gateway (when gatekeeper not used).

After the gateways determine that the VoIP call can be placed across the IP network, the gateways negotiate call capabilities using preconfigured coding profiles, and then optionally strip before sending the extension digits from the local to the remote gateway where they are delivered either to the phone, PBX, or keyset. The call can be processed as either a local or long distance call depending on how the remote gateway is configured. A ringing or busy signal is transmitted to the caller once the call is connected. If the call is answered, the gateway sends the voice or fax transmissions. If the wide area network is unavailable, calls may not go through, in which case callers receive a busy signal.

When a caller hangs up the receiver, the VoIP call session is terminated. Multiple gateway trunks may be used for all calls except those initiated from keysets which must go directly to the gateway.

Page 1-5

VoIP Telephone Calls

Local Telephone Number

Step

Source

IP Address

VoIP H.323 Gateway

Remote Telephone Number

VPN

Call setup completed

VoIP H.323 Gateway

PBX #1

IP call

VoIP Call Scenario — Step 3: Call Setup Completed

PBX #2

(message returned)

Destination

IP Address

Phone

Rings

Local Telephone Number

Step

Source

IP Address

VoIP H.323 Gateway

VoIP Call Scenario — Step 4: Remote Call Answered as VoIP Call

PBX #1

VPN

Call answered

VoIP H.323 Gateway

IP call

Remote Telephone Number

(message returned)

Destination

IP Address

PBX #2

Page 1-6

♦

Elements of a Converged Network

Alcatel’s H.323 VoIP gateway is based on a complex, dual-technology infrastructure taken from what have been in the past two fairly distinct industries — namely, Telecommunications (a.k.a. Telephony) and Data Communications. It converges voice and data into enterprise, Internet Service Providers (ISPs), and carrier networks to provide various levels of VoIP services using intelligent switches in order to generate long-term cost reductions for telephone services between sites.

The standard or key elements of a converged H.323 VoIP network are described below and shown in the illustration

Circuit-Switched PSTN

representation of all the various devices that may be used in a VoIP network and how they may interconnected. VoIP network interoperability is based on ITU H.323 network call control standards and multiple vocoder support. See also abbreviated

tion Union (ITU) Standards

By means of either digital or analog VoIP daughtercards installed in Alcatel switches, the basic elements required for providing enterprise H.323 VoIP gateways in packet-switched IP networks are readily accommodated, including the client, the gateway, and the gatekeeper as described.

Elements of Converged Voice/Data Packet-Switched VoIP Network and

on page 1-9. This illustration is intended to provide a sample, visual

International Telecommunica-

on page 1-17.

VoIP H.323 Client

The Client is the device initiating and/or receiving the call. This can be a standard telephone handset or some other H.323 VoIP-capable device in an IP network.

VoIP H.323 Gateway

Alcatel’s H.323 VoIP Gateway is the device used to make the transition from the packetized voice network to a circuit-switched network, e.g., PSTN, and back. Functionally, the enterprise VoIP gateway is comprised of voice to IP network converter components, .e.g, DSPs, on the voice switching daughtercards. In VoIP, the process for call placement is the same as in a service provider system except that the gateway is accessed from Customer Premise Equipment (CPE) instead of from a local service provider, e.g. CLEC (Certified Local Exchange Carrier).

Note ♦

PBX and Key Systems setup, installation and configuration procedures are beyond the scope of this manual.

Gateway devices intercept then direct electric signals between networked devices. With VoIP, gateways translate transmission formats between voice CPE and H.323 IP network call control endpoints and terminals, including communications procedures between gateways. They also translate between codecs, perform call setup/teardown on LANs and on circuit-switched telephone networks. Gateways are entrance and exit points into VoIP networks that without hardwiring perform code and protocol conversions, as well as signal filtering.

VoIP gateways contain a user-definable phone directory database of phone number to IP address mappings; this is called an Alcatel VoIP Network Dialing Scheme (AVNDS). See Chapter 3, “Network Dialing Schemes,” for details. Modifications to the local phone directory database are downloaded through the IP network to the switch, and may be accessed using the VoIP configuration interface. The phone directory database is built as the VoIP network is configured, and is contained in the VoIP configuration boot file ( plans, phone groups and destinations as part of the AVNDS comprise a portion of the phone directory database used by Alcatel’s H.323 VoIP gateway.

vsmboot.asc

). Numbering

Page 1-7

Elements of a Converged Network

Gateways are considered H.323 terminals or H.323 endpoints in H.323 IP networks. Terminals are also the endpoints where telephone lines connect to network circuits. Terminals provide real time, two-way communications for local area network (LAN) endpoint destinations. All terminals as such must support voice communications and H.245 in-band call controls to use and negotiate channels. See also abbreviated International Telecommunica-

tion Union (ITU) Standards

VoIP H.323 Gatekeeper (Optional)

The H.323 Gatekeeper (server or workstation) is the device that verifies client VoIP privileges and translates telephone numbers into IP addresses. It should be noted that H.323 gatekeepers are not required to use Alcatel’s H.323 VoIP Gateway. In lieu of an H.323 VoIP gatekeeper, Alcatel’s H.323 VoIP gateway uses its patent-pending Alcatel VoIP Network Dialing Scheme (AVNDS) to perform IP address translations.

Gatekeeper setup, installation and configuration procedures are beyond the scope of this manual.

on page 1-17.

Note ♦

♦

Alcatel recommends and has tested extensively use of Alcatel’s H.323 VoIP gateway with the NT100 RADVision Gatekeeper.

Gatekeeper devices identify, track and control traffic flowing through them, and perform other functions such as gateway registration, admission and bandwidth controls.

Page 1-8

VOICE

Gatekeeper

Elements of a Converged Network

LAN

Clients

Central Site

BRI

Telephone

PBX #1

4400

Ethernet

OmniPCX

H.245

VoIP H.323 Gateway

Digital

Packet-Switched

VoIP Network

WAN

Euro

ISDN

ISDN PSTN

Ethernet

VPN

WAN

VoIP H.323 Gateway

WAN

Circuit-Switched

NO. AMER.

PSTN

H.245

Microsoft

NetMeeting

IP Address

Remote Site

PSTN

Fallback

(Deadman

Relay Switch)

Client

Key System

POTS/ PSTN

Analog

VoIP H.323 Gateway

FXO/

Elements of Converged Voice/Data Packet-Switched VoIP Network

and Circuit-Switched PSTN

Page 1-9

H.323 VoIP Gateway Voice and Convergence Features

As shown below, the main functions handled by Alcatel’s H.323 VoIP Gateway include the following:

• Telephony Signaling

— used to communicate with the PSTN or Customer Premises

Equipment (CPE).

• Payload Packetization and Digital Signal Processing (via DSP)

— converts PCM voice packets from circuit-switched network to H.323 packets on IP network and the reverse.

• H.323 Network Call Control

— handles H.245 and H.225 packet processing, e.g.,

connect, disconnect.

• Alcatel VoIP Network Dialing Schemes (AVNDS)

— handles conversions between

phone numbers and IP address of H.323 devices.

• Network

Switch Backplane Interface — connects H.323 VoIP gateways to switch and

ultimately to IP network.

These functions can generally be divided into either voice or convergence features, based on the controls they provide over VoIP in the switch. For the most part, the voice features include separate controls for signaling and for voice interoperability, whereas, the convergence features encompass H.323 call control and voice/data interoperability via the use of AVNDS in IP networks.

Alarms

PSTN, PCX/PBX, BRI phone

Alarms

Signaling

Voice Ports

Payload

Telephony Signaling

(Digital or Analog)

DTMF

(Digit

Collector)

Daughtercard Activation

Payload Packetization

(Voice, Fax, Modem)

Configuration

Alcatel

Voice Network

Dialing Schemes

(AVNDS)

VoIP Network

Call Control

(H.323)

VoIP Daughtercards and Enterprise VoIP Features

Control Packets

Network

Switch

Backplane Interface

RTP Payload Packets

Switch

Bus

Page 1-10

H.323 VoIP Gateway Voice and Convergence Features

Signaling Control and Voice Interoperability (Voice Features)

The ability to accommodate voice traffic using VoIP switches installed in data networks is achieved by means of signaling controls and voice interoperability features. The VoIP signaling control and voice interoperability functions includes telephony signaling and payload packetization as described.

Telephony Signaling

Telephony signaling is used for signaling with telephone equipment, e.g., PBX, via the telephony interface, as well as to control the communication signaling between the H.323 VoIP gateway and the Customer Premises Equipment (CPE). It detects the presence of new calls, collects dialed digit information (telephone number in some form or another) entered by the caller to route a call via an AVNDS to its destination point, and is also used to detect the end of calls (off hook).

Telephony signaling provides call progress supervision by generating supervisory and call progress tones, as well as DTMF (Dual Tone Multi Frequency) tones for outbound calls. It also provides DSP (Digital Signal Processor) interfacing control, and transfer of PCM-based voice packets to and from the DSP subsystem or DIMM (DSP Interface Management Module). It coordinates with the DSPs to select voice coders (codecs) at startup when a particular vocoder is needed. When a call is received, telephony signaling is responsible for opening channels and PCM data streams to the DSPs to process the voice data.

The signaling controls provided by the Telephony Signaling functions includes the following:

• Call Progress Tone and Tone Detection converts them into tones or other signaling events, e.g., answer or busy signals.

• Dialing Timers wink start, or how long to wait for another digit.

• E&M Signaling (Common, Wink Start, Immediate Start and Delay Start) attribute settings or parameters to match CPE. (Available only on VSD-T1.)

• Foreign Exchange Station (FXS) Loop Start ters to match CPE.

• Foreign Exchange Office (FXO) Loop Start ters to match CPE.

• Caller ID between first and second ring.

The telephony signaling configuration options for telephone signaling interfaces, e.g., ring delay and cadence (ringing rhythm), are assigned to the physical ports on the daughtercard, including the T1 and E1 line specifications. All options are defined at the channel level.

Parameters for telephony signaling and VoIP network preferences are pre-configured in textbased configuration files referred to as boot files and subsequently assigned to a daughtercard and/or its components. For more details on setting these parameters, see also Chapter 5, “VoIP Commands.”

— used to time incoming signaling events, e.g., how long to wait for a

— looks for Caller ID information, e.g., calling party telephone number

vsmboot.asc

— detects individual in-band frequencies and

— customizes

— customizes attribute settings or parame-

files. The parameters are stored in the

Page 1-11

H.323 VoIP Gateway Voice and Convergence Features

Payload Packetization and Digital Signal Processing

Payload packetization is responsible for conversion between time-continuous telephony (analog or digital payload) at the telephony interface and Real Time Protocol (RTP) packets on the data network interface. It supports voice compression, echo cancellation, Fax and DTMF Relay (demodulation/modulation), modem data transport (up to 14400 baud), voice activity detection and comfort noise generation, as well as packet arrival de-jittering.

Physically, the payload packetization function is implemented on the DSPs (DIMMs), with control and configuration on the Motorola MPC860 processor. Configuration is performed through the

vsmboot.asc

file on the switch. Upon VoIP daughtercard activation, the configuration is transferred from the switch to the daughtercard. See Chapter 4, “Setup and Installation,” for more information.

The controls for voice interoperability provided by the payload packetization functions include the following:

• Codecs

(see also Coding Profiles -- H.323 Call Capabilities) — provides encoding/

decoding of H.323 packets.

• Voice Echo Cancellers

• Fax or Modem over IP

• Voice Activity and

— reduces echo on voice conversations.

— allows fax/modem calls to be transmitted via H.323.

Silence Detection — detects voice conversation (or lack thereof) to

reduce H.323 bandwidth requirements.

• Comfort Noise and Jitter Buffer

— generates slight background noise (white noise) on

the voice conversation, so callers do not think the connection has failed.

Digital Signal Processors, or DSPs as they are more commonly known, are math-intensive coprocessors used to convert and manipulate information, especially in telecommunications systems (systems that transmit all types of data including voice and video). They are also programmable chips, well-suited for VoIP as DSPs have the ability not only to convert but to compress analog signals into various digital formats, i.e., perform digital signal processing.

Although DSPs do not have any direct analog input/output since they are actually digital devices, they can accept digitized analog data rather than raw analog signals. As a result, DSPs are used in the digital and analog VoIP daughtercards developed by Alcatel to bring switch-enabled VoIP to enterprises; however, before the digitized and compressed voice signals can be delivered as “voice data” in a VoIP network, they must be packetized into H.323 packets.

Packetized voice is digitized voice compressed into finite bit stream of IP packets, that carry the “voice payload” between remote and distant locations, across the IP network and make processing VoIP calls in IP networks possible. Once compressed and packetized, periodic delays (jitter) to make the call sound smoother must be imposed on the transmission of these packetized “voice” conversations to mimic “real time voice” (resonating by nature in continuous “analog” waveform). DSPs are used further to reduce the delays from conversion and compression to ensure quality voice communications without affecting the real time voice processing and compression that occurs simultaneously.

To transmit the compressed data (digitized voice) across the IP network, the Real Time Protocol (RTP) is used. RTP streamlines and then transports voice packets, including interactive multimedia packets over IP, although it does so without any guarantees or quality of service provisioning.

♦

Note

♦

H.323 VoIP telephone calls automatically receive the highest priority in the VoIP network via the Quality of Service ToS bit. For more information, see the switch manual.

Page 1-12

H.323 VoIP Gateway Voice and Convergence Features

Voice packet transmissions, or the “payload,” are expedited by engaging the User Datagram Protocol (UDP) for faster delivery, packets which by necessity include the IP network call transport header information. Resultant jitter caused by delays imposed on the payload packets upon arrival to their destinations is also handled by the DSPs.

Layer 2 Header

UDP is needed by RTP to keep pace with “Real Time Voice” but lacks controls and error checking capability.

DSPs can monitor calls in progress, detect voice activity and handle echo cancellation (the filtering of unwanted transmission signals as specified in ITU algorithm standards G.160 and G.126); comfort level background (white) noise can also be generated on either the transmitting or receiving end.

Since digital signal processing affects nearly every operation in VoIP, numerous DSPs are incorporated adjacent to the supporting MPC860 CPU signaling controller in the voice switching daughtercards (normally used with voice switching modules), comprising the core of Alcatel’s enterprise VoIP on the call processing end. The DSPs and the Motorola MPC860 controlling processor work in unison to support the various protocols and interfaces that implement the enterprise VoIP telephony functions contained in software on the voice switching daughtercards. In a nutshell, the DSPs are the voice processors, and the MPC860 controller is the data communications processor on the daughtercards. Altogether, the above components provide T1, E1 and ISDN voice and data synthesis processing, with scalable versions of each bringing enterprises any-to-any switching functionality that now, with enterprise VoIP, includes least-cost call routing for VoIP Virtual Private Networks (VPNs).

Header

UDP

Header

Voice Packet Transmission

RTP

Header

Voice/Fax

Data Payload

Signal Recognition

Initially, digital signal processing involves DSP detection of an array of voice signaling types using Channel Associated Signaling (CAS) repetitive circuit-state signaling protocols (for T1 and E1 lines). Many forms of call signaling exist to set up and end calls, most of which result in the ringing of a phone or connection of a fax machine. These forms entail newer line signaling methods that use digital pulses (PCM, or Pulse Code Modulation), analog touchtones such as DTMF (Dual Tone Multiple Frequency), and other much older analog signals in all their assortments, including but not limited to: Ear & Mouth (E&M), Loop Start, Ground Start, Foreign Exchange Subscriber (FXS) and Wink Start. Each signaling method was developed through the years by the telephone industry to provide Plain Old Telephone Service (POTS).

E&M signaling, of which there are five interface types, is the most widely used method for connecting calls to PBXs, telephone switching systems which use channelized T1 or E1 lines to transmit signals and multiplex digitized voice. T1 robbed bit signaling is an example of narrow or in-band signaling — where signaling tones are passed along the same circuit as someone’s voice.

ISDN (Integrated Services Digital Network), on the other hand, is another type of signaling wherein voice transmissions are digitized then placed on separate broad or out-of-band channels (so signaling tones are not passed along the same circuit as someone’s voice). This prevents signaling or other intrusions into the calls, and usually provides faster transmission.

ISDN is a common protocol in the Common Channel Signaling (CCS) network architecture used for exchanging information between out-of-band signaling networks and telecommuni-

Page 1-13

H.323 VoIP Gateway Voice and Convergence Features

cations nodes in the network. ISDN does not use T1 (or DS-1) robbed bit signaling, where bits are taken from voice data to carry signaling. Alcatel H.323 gateways support in-band and out-of-band signaling.

Encoding

Once signaling types are determined they are analyzed and converted by the appropriate DSP voice coder (vocoder) into digital signals, which are ultimately converted and expanded back (re-modulated) into real voice. More specifically, after signal recognition and analysis, DSPs convert (encode) the amplitude of incoming analog signals into digital form using codecs, or CODer/DECoders.

The basic encoding schemes, or companding methods in use today, are for PCM which “encodes” analog signals into digital signals. Although the PCM companding methods used for T1, which follows Mu-Law, and E1 which follows A-Law, differ mainly in their algorithms, their purpose is much the same. (Companding is a contraction for compression and expansion.) However, A-Law and Mu-Law are incompatible. They use different methods, for example, to sample analog signals.

Next, the digitally encoded signals are compressed using industry standard vocoders. These are devices that use speech compression/decompression algorithms to analyze and convert analog waveforms into digital signals and reduce related bandwidth requirements.

Vocoder

G.711 G.723.1 G.729 G.729a

Compressed

Real Time

Protocol

Encapsulation

Analog

DSP

Digitizer

Digital

Voice/Data Encoding and Call Compression

Compression

The appropriate vocoder used for VoIP calls is then negotiated by the H.323 VoIP gateway prior to call placement. As an added bonus, but with some variations in protocols, the same DSP technology that is used for voice compression also works with fax modems. (For that reason, it can be assumed that references to voice signal packets inherently include “fax” packets.)

The codecs and vocoders used in enterprise VoIP adhere to the ITU recommendations that fall under the H.323 IP network call control umbrella of interoperability standards for multimedia communications over packet-switched local area networks (part of the Series H Recommendations for Audiovisual and Multimedia Systems). The ITU’s H.323 suite of specifications includes the H.245 in-band call control specifications. For the signaling vocoders (G.711, PCM; G.723.1, Internet Speech; G.729/G.729a; Standard Telephone Speech), the algorithms in the Series G Recommendations for Transmission Systems and Media, Digital Systems and Networks are used.

Page 1-14

H.323 VoIP Gateway Voice and Convergence Features

H.323 Call Control and Network Interoperability (Convergence Features)

The ability to accommodate voice (also fax and modem) traffic compressed into data form via payload packetization for transport across data networks is achieved through the use of H.323 call controls and Alcatel VoIP Network Dialing Schemes (AVNDS) as described.

H.323 Network Call Control

H.323 network call controls are responsible for the procedures and protocols necessary to establish/tear down VoIP calls across the IP network. The VoIP gateway implements the H.323 network call control standards, which include the following:

• The H.225/Q.931 protocol that performs call establishment and tear down by establishing a reliable call signaling channel.

• The H.245 protocol that establishes a reliable H.245 in-band channel for communications between all endpoints or terminals, i.e., gateways, for capability exchange and other messages.

• The registration, admission and status (RAS) protocol that creates a RAS channel to carry RAS messages between an endpoint and a gatekeeper.

• The H.323 IP network call control standards that support multimedia communications over local area networks.

The H.323 network call controls and capability provided by the H.323 call control functions includes the following:

• H.323 gateway (VoIP Switch)

• H.323 gatekeeper (e.g., RADVision Server) (Discovery, Configuration and Operation)

• H.323 call capabilities (Coding Profiles or Codecs, Voice Network Delay Buffers)

The voice network configuration options include general network information, H.323, H.225, and H.245 configuration settings, gateway, gatekeeper and registration parameters, and Real Time Transport Protocol (RTP) session parameters that must be specified for IP network communications. Voice network call control parameters are configured at the VoIP daughtercard level.

Alcatel VoIP Network Dialing Schemes (AVNDS)

The AVNDS are responsible for the operations and configuration of the VoIP daughtercard and/or voice switching module, e.g, VSX. AVNDS are implemented on the Motorola MPC860 processor, and the switch.

AVNDS are responsible for providing the interface to configure and maintain all VoIP daughtercards (H.323 gateways) on the entire VoIP network. Additionally, both standard packet Management Information Bases (MIBs) and proprietary voice packet MIBs are supported.

The AVNDS are used to store information contained in VSM (Voice Switching Module) configuration boot files ( larly the following:

vsmboot.asc) concerning the configuration of the VoIP network, particu-

• Destinations (H.323 endpoints, H.323 local channel destinations)

• Phone Groups (e.g., strip digits and extensions)

• Numbering Plans (hunt methods and hunt groups)

Page 1-15

H.323 VoIP Gateway Voice and Convergence Features

The AVNDS handle inbound/outbound calls routing to/from the VoIP network and local ports. AVNDS are also used to set up calls and translate IP addresses to telephone numbers, and can be used with, or in lieu of, H.323 VoIP gatekeepers.

VoIP configuration boot files and profiles simplify VoIP configuration of Alcatel’s H.323 VoIP gateways (VoIP daughtercards) by using sets of pre-configured parameters that can be assigned to the various manageable components. Various configuration elements, e.g., profiles, have a user-defined name associated with it. VoIP daughtercard configurations are stored in the switch.

Destinations which consist of remote network and local calling gateways, including H.323 gatekeepers, allow Network Administrators to configure a destination IP address and its specific protocol. Local channel destinations are considered subdestinations. Destinations, which are appended to hunt methods, are configured at the daughtercard level.

Phone groups are used to indicate what telephone numbers are available. They also define digits to be stripped and forwarded. Phone groups are configured at the daughtercard level.

Voice numbering plans use hunt methods to arrange telephone lines so that when calls come into the network they will ring in a certain order. For example, to use PSTN fallback, all phone groups must be set up with the last group element indicating the local destination or gateway to fall back on when a call cannot be placed over the VoIP network.

Hunt methods in voice numbering plans are configured at the daughtercard level. Hunt methods dictate what to do if the first line tried is busy, i.e., hunt methods are used to track down lines in a certain order until an available line is located. Phone line destinations can be grouped as desired in user-defined groups, such as by divisions or departments, location, or some other meaningful grouping.

For information on setting up and using the AVNDS (Alcatel VoIP Network Dialing Schemes), see Chapter 3, “Network Dialing Schemes,” Chapter 4, “Setup and Installation,” and Chapter 5, “VoIP Commands.”

Switch Backplane Interface

The switch backplane interface is responsible for the payload packet transport, and VoIP daughtercard management message transport between the VoIP daughtercard and the host switch. Physically, the interface consists of a 100-pin connector between the VoIP daughtercard and the motherboard. All functions of the H.323 VoIP gateway are implemented on the MPC860 controllers on the daughtercards, the VSX (OSR configurations only) and the switch.

Page 1-16

VoIP Standards for Development

Alcatel’s H.323 VoIP gateway is designed to function in accordance with the following IP Telephony and Internetworking standards currently available as briefly summarized below.

International Telecommunication Union (ITU) Standards

ITU-T technical, operational, and tariff recommendations are used for standardizing telecommunications on a worldwide basis. ITU H.323 IP network call control standards apply to VoIP. These standards define the major components, namely, Telephone Terminal Equipment, Gateways, Gatekeepers and Multipoint Control Units (MCUs) for H.323–based communications systems:

Series H.323 — Series H: Audiovisual and Multimedia Systems (Infrastructure of audiovisual services — systems and terminal equipment for audiovisual services). H.323: this standard is specifically concerned with recommendations for real time audio, video and/or data and facsimile transmissions over H.323, Packet–based Multimedia Communications Systems. In reference to Alcatel’s enterprise VoIP, the series as a whole relates to gateway devices used in VoIP to handle audio, video, data and facsimile transmission over IP or packet networks. The standard specifically includes the newer ITU recommendations for the Internet facsimile protocol (T.38) that is used to 1) exchange messages and data between facsimile gateways connected via an IP network, and 2) message transport (depending upon bandwidth availability) using either TCP/IP or UDP/IP network protocols. (T.38 is incorporated into the first release of Alcatel’s enterprise VoIP).

Series H.225 — Series H: Transmission of Non–Telephone Signals (Infrastructure of audiovisual services — Transmission multiplexing and synchronization. H.225: this standard is specifically concerned with recommendations for narrowband visual telephone services defined in H.200/AV.120-Series transmission paths for local area networks (LANs) providing non-guaranteed quality of service (QoS) which is less than that of ISDN PRI protection and recovery mechanisms. This recommendation describes how non-guaranteed QoS LANs provide conversational services for audio, video, data and control information in H.323 equipment. The series relates to gatekeeper devices used in VoIP to provide services across LANs. Gatekeepers are centralized network devices performing IP address translations and bandwidth management. Alcatel’s H.323 VoIP gateway uses AVNDS to work with, and in lieu of, gatekeepers. (Note: Standard includes codec support for call synchronization.)

Series H.245 — Series H: Audiovisual and Multimedia Systems (Infrastructure of audiovisual services — Communications procedures), Control protocol from multimedia communication. H.245: this standard specifies syntax and semantics of terminal information messages, particularly receiving and transmitting capabilities, mode preferences from the receiver, including logical channel signaling, Control and Indication messages. Signaling acknowledgements are specified to ensure reliable audiovisual and data communications. The series relates to Multipoint Control Units used in VoIP to provide signaling and coding for call synchronization.

Codec Support (G.711, G.723.1, G.729a)

G.711 (PCM Encoding)

This is the ITU recommendation for an algorithm designed to transmit and receive A-Law and Mu-Law PCM voice at digital bit rates of 48, 56, and 64 Kbps. It applies to digital telephone sets on digital PBS (cellular) and ISDN channels. Support for this algorithm is required for ITU-T compliant videoconferencing (the H.320/H.323 standard).

Page 1-17

VoIP Standards for Development

A-Law and Mu-Law are processes needed to compand digital signals. A-Law is used in most countries except for the U.S., Canada and Japan where Mu-Law is more common. Companding is the process of compressing the amplitude range of a single signal, and then expanding them at the receiving end back to their original form. Although it is impossible to exactly reproduce an analog signal digitally, companding greatly improves the accuracy of this process. PCM uses two different companding processes. For this reason, PCM A-Law is used for international networks.

A-Law: The PCM coding and companding standard used in Europe and in areas outside of North America. A-Law encoding samples audio waveforms used in the 2.048 Mbps, 30-channel PCM system (E-carrier).

Mu-Law (E-Law): The PCM voice coding and companding standard used in Japan and North America. A PCM encoding algorithm where analog voice signals are sampled 8,000 times per second with each sample represented by an eight-bit value, and a raw 64 Kbps transmission rate. All sample bits are inverted before transmission.

A-Law and Mu-Law are incompatible. For example, a signal sent with A-law cannot be received by a system using Mu-Law.

G.723.1

♦ Note ♦

This is the ITU-T algorithm recommendation used for compressed digital audio over Plain Old Telephone Service (POTS) lines. It is the voice part of H.324 (POTS video conferencing). This algorithm runs at 6.3 or 5.3 kbps (20 bytes per 30ms interval) and uses linear predictive coding and dictionaries, which help provide smoothing. The smoothing process is CPU-intensive during real time based activities.

G.729a

This is the ITU’s standard voice algorithm – CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Predictive for the encoding/decoding of speech at 8 Kbps using conjugatestructure, algebraic-code excited linear predictive method. G.729 is supported by inter alia (among other things), American Telephone and Telegraph, France Telecom and Japan’s Nippon Telephone and Telegraph.

VON (Voice on the Net) Developments

The VON (Voice on the Net) Coalition is concerned with developments in Internet Telephony and IP Telephony around the world. It is an incorporated, non-profit U.S. organization working with the government, business and other groups and individuals on regulations that affect this technology and its use. Alcatel’s H.323 VoIP gateway was designed with the considerations of the VON Coalition in mind. Compression techniques and DSPs improve the quality of VON transmissions and minimize problems associated with IP packet delays.

Page 1-18

VoIP and VLANs

Alcatel VoIP (VSD, VSB, VSA) modules cannot not be in a Virtual LAN (VLAN) with non-voice ports (i.e. data ports), IP phone ports, etc. All voice traffic must route in and out of the VoIP

VLAN.

VoIP and VLANs

Page 1-19

VoIP and VLANs

Page 1-20

2 VoIP Daughtercards

Introduction

This chapter describes the voice switching daughtercards that can be installed in Alcatel switches to provide H.323 VoIP gateways in VoIP networks. Using ITU H.323 IP telephony standards, the H.323 VoIP gateway converts telephone or fax calls between the circuit switched Public Switched Telephone Network (PSTN) and packet-switched VoIP networks. Alcatel’s H.323 VoIP gateways are typically used to handle VoIP calls as such placed across local and wide area networks between branch offices in remote enterprises, although the gateways are suitable for use in carrier applications, too. See Chapter 1, “VoIP Overview” for a more in-depth description of Alcatel’s VoIP H.323 gateway operations.

Different VoIP daughtercards, as described below, are required depending on the telephony interface required to transmit and receive calls in the VoIP network. Furthermore, to digitize the VoIP calls, the daughtercards utilize digital signal processors (DSPs) containing a specified number of channels, which in turn determine the maximum number of calls that can be placed at one time on the card.

The VoIP daughtercards are referred to as voice switching daughtercards and, when installed in the switch, they are sometimes referred to as Voice Switching Modules (VSMs). Currently, the VoIP daughtercards can be installed in either the OmniAccess 512 or Omni Switch/Router. A blade installed in an OSR containing either one or two VoIP daughtercards of the same type is referred to specifically as a VSX or VSX switching module (for details see VSX Switching Module on page 2-25). For information on configuring either of these switches, refer to the appropriate switch user manual.

This chapter also depicts the port pinouts and jumper settings for all of the VoIP daughtercards where necessary, as well as the Deadman switch, and Cross-Over toggle switches available on certain voice daughtercards. The front panels for the VoIP daughtercards, including the front and bottom views of the cards, are shown to illustrate certain components relative to important operations of the H.323 VoIP gateway in the switch.

All VoIP daughtercards can be field-installed. For details on installing the cards, see also Chapter 4, “Setup and Installation.” For details on configuring the switch to run VoIP, see Chapter 3, “Network Dialing Schemes” and Chapter 5, “VoIP Commands.”

Page 2-1

Introduction

VoIP Daughtercard Types

There are two types of VoIP daughtercards: digital and analog. The digital voice switching daughtercards includes the VSD daughtercard used for digital calls placed in either North America and/or Europe, and the VSB daughtercard used specifically for digital calls placed in Europe. The analog voice switching daughtercard (VSA) is used only in North America for placing analog POTS (Plain Old Telephone Service) calls, e.g., to the PSTN. The basic VoIP daughtercards, which allow the switch to make these various types of phone connections, are listed and described below.

• VSDs (T1, or E1 QSIG and E1 ISDN PRI Digital) (North America and Europe)

• VSBs (Euro BRI ISDN Digital) (Europe)

• VSAs (Analog) (North America and Europe)

VSD — The digital voice switching daughtercards (VSDs) have two physical port connections which can be either T1 or E1 (called Dual T1 or Dual E1). Associated with each of the digital physical ports there can be either 48 channels for T1 connections, or 60 channels for E1 connections. The VSD card supports the following protocols or voice port interface connections for VoIP networks in North America: T1. The VSD card supports the following protocols or voice port interface connections for VoIP networks in Europe: E1 (QSIG) or E1 ISDN PRI (QSIG is another name for ITU Q.931). The VSD card does not support the following protocols: T1 ISDN PRI, T1 QSIG or Euro BRI ISDN (E1 ETSI). The VSD-60CH T1/E1 card is considered a high-end VoIP daughtercard as it provides the most channels. Reliable non-digitized voice processing is available only between two ports of the same interface type on a single daughtercard, and not between daughtercards. See Voice Switching Daughtercard — Digital on page 2-6 for more details. For more information on the voice port interface types for the digital VoIP daughtercards see also the digital port configuration commands in Chapter 5, “VoIP Commands.”

VSB — The other type of digital voice switching daughtercard is normally referred to as a VSB since it provides Euro BRI ISDN (E1 ETSI) protocol or interface port connections for VoIP networks in Europe. It differs mainly in that it has four ports, each with two (B) bearer channels and one (D) data channel. B-channels carry voice and data content, whereas D-channels are dedicated to carry control signals or call processing data for the B-channels. Each B-channel contains one DS0 voice channel. The VSB card does not support the following protocols: T1 ISDN PRI or T1 QSIG; T1 or E1 QSIG; E1 PRI or T1 BRI. See Voice Switching Daughtercard — Euro BRI ISDN on page 2-13 for more details.

VSA — The analog voice switching daughtercard (VSA) can contain an even number of analog ports from two to 16 depending on whether the card provides Foreign Exchange Station (FXS), e.g., telephone set (TelSet), or Foreign Exchange Office (FXO), e.g., Central Office (CO) port interface connections in an OmniAccess 512 or Omni Switch/Router. Each analog FXS port allows the connection of one off-the-shelf TelSet, or some other voice device, e.g., analog fax machine, analog phone answering machine, whereas each analog FXO port allows the connection of an FXO cable to a wall outlet or CO). SeeVoice Switching Daughtercard — Analog on page 2-19 for details.

♦ Note ♦

When used separately, the terms E1 and ETSI both entail European PRI and BRI interfaces. E1 ETSI used together as one term refers specifically to Euro PRI.

Page 2-2

Introduction

The table below shows the basic versions of the VoIP daughtercards, all of which were designed with various configurations in mind to fully support the wide range of features used in Voice over IP. See also Chapter 4, “Network Dialing Schemes,” for more configuration details. Also, not all configurations shown below may be currently available for purchase.

Voice Daughtercard Description

VSD VSD-12CH Two (2) digital T1/E1 RJ-45 voice ports, 12 compressed voice channels

VSD-24CH Two (2) digital T1/E1 RJ-45 voice ports, 24 compressed voice channels

VSD-36CH Two (2) digital T1/E1 RJ-45 voice ports, 36 compressed voice channels

VSD-48CH Two (2) digital T1/E1 RJ-45 voice ports, 48 compressed voice channels

VSD-60CH Two (2) digital T1/E1 RJ-45 voice ports, 60 compressed voice channels

VSB VSB Four (4) digital E1 (Euro BRI) RJ-45 voice ports, TE (Terminal Equipment), NT

(Network Terminator), Point to Point or Point to Multipoint NT, 8 compressed voice channels

VSA-FXS VSA-4FXS Four (4) analog RJ-11 voice ports

VSA-8FXS Eight (8) analog RJ-11 voice ports

VSA-FXO VSA-2FXO Two (2) analog RJ-11 voice ports

VSA-4FXO Four (4) analog RJ-11 voice ports

VSA-MIX VSA-4FXS-

2FXO

Four (4) analog FXS and (2) analog FXO RJ-11 voice ports

Page 2-3

Introduction

Digital Signal Processors (DSPs), DIMMs and Available Channels

All digital VoIP daughtercards have four vocoder channels available per DSP chip. DSPs are scalable in increments of three (in DSP DIMM modules) to better accommodate the needs of a VoIP network, and to reduce costs since the number of DSPs required is based on the number of simultaneous vocoder channels needed. DIMM stands for DSP Interface Management Module (not Dual Inline Memory Module).

Voice switching daughtercards, such as the VSD T1/E1 card, can contain up to 15 digital signal processors running at 100 MIPS (millions of instructions per second), providing voice processing functions for up to 60 DS-0s, or 60 Digital Service 0 channels (24 DS-0s are equal to one DS-1, or T1, channel).

The number of populated DIMMs (DSPs) on the digital voice daughtercards determines how many channels, which are bidirectional, are available on the card (e.g., 12 bidirectional channels per DIMM; or, a maximum of 60 bidirectional channels with 4 additional DIMMS). The number of simultaneous channels available on a particular digital voice switching daughtercard can be determined easily by counting the DIMMs on the card.

The illustration on the next page shows a VSD T1/E1 card fully-populated with standard DSPs and additional DIMMs. As a minimum configuration, three DSPs come standard on the digital cards to provide 12 channels (4 bidirectional channels per DSP); however, it is strongly recommended that only fully-populated (128 MB, 60 channel) VSD daughtercards be installed for the following reasons.

• Although a VSD with 12 channels (three std. DSPs) has two operational T1 ports that can provide up to 48 bidirectional channels, only the first 12 simultaneous calls can be handled per DSP; therefore, without additional DSPs (DIMMs) the 13th call and all subsequent calls will be ignored completely, i.e., no dial, busy signal or comfort noise will be generated, until either a channel becomes available, or additional DIMMs are installed.

• The installation of additional DIMMs, in effect, provides redundancy in the event of a DSP failure.

♦ Notes ♦

There are no DIMMs on the VSA daughtercard per se, only on the FXO or FXS grand-daughtercards with a maximum of one channel each per port. See Voice Switching Daughtercard — Analog on page 2-19 for more details.

DIMMs are not field upgradeable; however, the flash memory on the boards is field upgradeable. The flash memory must always match the image used or the daughtercard will not function properly if at all. Contact Alcatel’s Customer Support for details on obtaining the appropriate flash and/or corresponding image upgrade.

Page 2-4

Introduction

Digital

Voice Port

Digital

Voice Port

Digital

Voice Port

(4 DIMMs =

48 Channels)

12 Channels

DSP

M M

DSP

M M

DSP

Standard DSPs

DSP

Top View

I M M

Switch

Bus

DSP

M M

DSP

Bottom View

Switch

Bus

Digital

Voice Port

flash

Digital Voice Switching Daughtercard (VSD T1/E1) — DSPs/DIMMS Top and Bottom Views

Page 2-5

Voice Switching Daughtercard — Digital

The digital Voice Switching Daughtercard (VSD) is used to provide digital telephone connections in Alcatel’s H.323 VoIP gateways. There are two main types of digital voice switching daughtercards (VSDs) that can be used to provide VoIP: North American T1 or European E1 (QSIG or Euro ISDN PRI) and (VSBs) Euro BRI ISDN (E1 ETSI). Euro BRI ISDN; see also Voice Switching Daughtercard — Euro BRI ISDN on page 2-13. Each VSD contains two ports per daughtercard and up to 24 DS0 channels (T1) or 60 DS0 channels (E1) per port. A maximum of one daughtercard can be installed per OA-512 switch (see VSD Front Panel on page 2-7), and up to two daughtercards can be installed in a VSX in each available slot of an Omni Switch/Router.

All in all, there are five main daughtercard DSP/DIMM configurations on the VSD version of the digital cards:

• 12 channels (0 DIMM; only standard DSPs)

• 24 channels (1 DIMMs)

• 36 channels (2 DIMMs)

• 48 channels (3 DIMMS)

• 60 channels (4 DIMMS)

Each VSB contains four ports per daughtercard and two ISDN BRI B-channels and one Dchannel per port. A maximum of one daughtercard can be installed per OA 512 switch (see VSB Front Panel on page 2-15), and up to two daughtercards can be installed in a VSX in each slot of an Omni Switch/Router. The VSB supports eight channels via two standard DSPs with four channels each, but does not support any add-on DIMM modules.

The OmniAccess 512 chassis provides one empty expansion slot (labeled as use with features such as Voice Over IP (VoIP); it does not accept the VSX switching module used in Omni Switch/Routers. VSDs, VSBs and VSAs cannot be installed in the same slot in an OSR, and an MPX card is required in the OSR. See VSX Switching Module on page 2-25 for more information. Port numbers can vary depending on the VoIP switch configuration; see also VoIP Daughtercard Port Numbering Schemes on page 2-28.

All VSD ports are digital 8-pin, RJ-45 voice ports containing from one to 60 channels per port, depending on whether the voice port interface type is T1 or E1.

All VSD and VSB daughtercards require 32 MB Flash memory, and for OSR configurations, 64 MB DRAM memory on the MPX. For power requirements, see VSX Switching Module on page 2-25.

The following FCC Class B certifications for the VSD daughtercards have been obtained to date: OA-512-VSD-36T1, -36E1, -48T1, -48E1 and -60E1.

♦ Notes ♦

The number of simultaneous calls per card is dependent upon the number of available DSP channels. Two channels are used per call, e.g., with 12 channels six simultaneous calls can be connected.

Calls between channels on the same VSD or VSB card use PCM (Pulse Code Modulation) instead of the H.323 protocol to process digital calls, and require two DSP channels to make the calls.

S4) reserved for

Page 2-6

Voice Switching Daughtercard — Digital

VSD Front Panel

Each port has three corresponding LED indicators with link status displays as shown below.

Reset Button: Insert

All VSD daughtercards have three LED displays per voice port as follows:

FAIL: On when VSD fails or diagnostic test fails, or when VSD image download fails.

Off when VSD hardware is functional, or when VSD image download is OK.

VSD (Voice Switching Digital) T1 or E1 Daughtercard Front Panels

pin to reset VSD. (Not available this release.)

VSD

A FAIL ERR LINK

ERR: On when T1 or E1 VSD voice port port link error occurs in line. This can be any T1 or E1 type of error, e.g., out-offrame/loss of synchronization.

LINK: On when T1 or E1 VSD voice port link to switch is connected.

Off when signal is lost and T1 or E1 VSD voice port link disconnected.

Page 2-7

Voice Switching Daughtercard — Digital

VSD Deadman Switch

The two types of digital voice switching daughtercards (VSDs and VSBs) contain mechanical relay switches referred to as “Deadman” switches. The Deadman switch is a relay switch that allows two telephony ports using standard telephone connectors, such as the 8-pin RJ-45 jacks used for transmission lines, to be connected to each other in the event of a power failure until 1) power is re-applied to the daughtercard, and 2) the switch reboots to break the Deadman connection and allows VoIP calls to again be placed. For more information on the RJ-45 jacks, see VSD RJ-45 Specifications on page 2-11.

The Deadman switch, which resets after 200 ms, also contains a watchdog timer. Because the timer keeps the switch relays open when the Deadman switch disconnects the RJ-45s from each other, they can be connected immediately to the framers terminating the digital or analog telephone line. So, if power is lost to the VSD (or VSB), the Deadman switch keeps the PSTN connection alive by connecting the two telephony ports together before the signals reach the framers. In other words, for all new incoming calls, a connection is maintained between one port connected to a customer’s PBX and the second port connected to the PSTN, otherwise known as PSTN fallback.

No special configuration is required to use the Deadman switch on the VSD T1 cards; however, to use it on the VSD E1 cards, one port on the card must be set to be the and the other port must be set to be the qslave.

On VSD E1 cards, setting both voice ports to (or both ports to qslave) will cause the two telephony switches to get alarms when the Deadman relay switch connects the two ports together. See the digital port configuration commands in Chapter 5, “VoIP Commands,” specifically the voice port isdn protocol command used to control the QSIG protocol settings.

qmaster

♦ Note ♦

qmaster

Page 2-8

PBX #1

PSTN

Digital

Voice Port

Digital

Voice Port

Deadman

Switch

Normally Open (disconnected) when power is re-applied to card.

Deadman Switch — PSTN Fallback Call Protection

Switch

Bus

Voice Switching Daughtercard — Digital

VSD Cross-Over Toggle Switch

The Cross-Over toggle switch for digital voice daughtercards (VSDs only) can be used to correct communication link errors between a daughtercard in the switch and a PBX or keyset due to the transmit (TX) and receive (RX) pins of the cable connecting the VoIP daughtercard and the digital telephony device (PBX or other voice device). This will show up as the link LED not turning green (see VSD Front Panel on page 2-7).

If a communication link error occurs between the switch and the PBX or Key Set as such, the blue Cross-Over toggle switch, as shown here on the top side of the board, can be flipped to a Cross-Over ON or OFF position after shutting down the VoIP switch and removing the affected daughtercard. This will swap the transmit and receive connections for the designated port. The default toggle position is to the left or OFF position.

Once the toggle switch has been flipped, the card can be reinstalled, and the ports on the voice daughtercard reconnected to the PBX or other voice device using either a StraightThrough or Cross-Over cable (Straight-Through recommended).

♦ Notes ♦

An amber cellophane tape may need to be peeled off the top of the Cross-Over toggle switch before the toggle switch can be flipped. When the tape is present it indicates the toggle switch is set to factory default.

Digital

Voice Port

Digital

Voice Port

The physical port always has 8 pins, but changes functionally depending on the cable in use. For more information on the RJ-45 jacks, see VSD RJ-45 Specifications on page 2-11.

Straight-Through

Default

Setting to Left

Cross-Over

Switches

(T1)

DSP

I M M

DSP

D I

DSP

M M

DSP

D I M M

DSP

M M

DSP

Top View

Switch

Bus

Cross-Over Switches — Swapping Port Transmit/Receive Connections

Page 2-9

Voice Switching Daughtercard — Digital

Cabling

Of the common cable types (compatible with RJ-45 jacks) that can be used with VoIP switches, the Straight-Through (Ethernet) and Straight-Through (T1 Voice) are both acceptable, as well as the Cross-Over (T1) cable; however, due to the TX/RX pinout wiring, the Cross-Over (Ethernet) cable cannot be used with VoIP switches. If the Cross-Over (Ethernet) cable is used the LINK LED will not display.

For E1 configurations, it is recommended that a balun connector always be used to connect a voice device (e.g., PBX) that uses an ITU G.703 interface (coaxial cables, BNC connectors) to a VSD (or any RJ-45 E1 port).

The balun converts the impedance of 120 Ohms on the RJ-45 port to 75 Ohms (G.723). The balun connector is not required when both ends have RJ-45 connections. Contact Alcatel’s customer support for more details on balun connectors.

♦ Note ♦

Page 2-10

Voice Switching Daughtercard — Digital

VSD Pinouts

The following illustration shows the pinouts for the digital voice switching daughtercard (VSD) 8-pin, RJ-45 jacks used to connect the voice ports on the card to voice devices in the VoIP network that support digital connections, e.g., PBX and Key Set.

♦ Note ♦

The pinouts as shown indicate when the Cross-Over toggle switch is ON and OFF.

VSD RJ-45 Speciﬁcations

Pin Number Standard Signal Name

1 Receive Data +

OFF

Cross-Over

Toggle Switch

2 Receive Data -

4 Transmit Data +

5 Transmit Data -

VSD RJ-45 Speciﬁcations

Pin Number Standard Signal Name

1 Transmit Data +

2 Transmit Data-

Cross-Over

Toggle Switch

4 Receive Data +

5 Receive Data -

Page 2-11

Voice Switching Daughtercard — Digital

VSD Jumpers

The following jumpers are factory set on the VSD daughtercard and should not be changed by the customer unless under the direction of Customer Support. Note that, in general, only jumpers which can be set with shunts, or are associated with ports on the board are identified and described.

This information is being provided solely for the purpose of repositioning a shunt which may have been inadvertently removed, so as to prevent damage to the board, and/or possibly render the board or other components connected to the VSD inoperable.

Jumper No. Shunt Position Default Port Description

P3 no shunt (on Pins 1_2) yes B RJ-45 connection

P3 no shunt (on Pins 3_4) yes A RJ-45 connection

P7 no shunt (on Pins 1_2) yes B RJ-45 connection

P7 no shunt (on Pins 3_4) yes B RJ-45 connection

♦ Caution ♦

P7 no shunt (on Pins 5_6) yes A RJ-45 connection

P7 no shunt (on Pins 7_8) yes A RJ-45 connection

P10 Pins 1_2 yes B RJ-45 connection

P10 Pins 3_4 yes A RJ-45 connection

Other VSD Jumpers

Jumper No. Shunt Position Default Description

P17 Pins 2_3 yes Backplane interface (Hbus)

Page 2-12

Voice Switching Daughtercard — Euro BRI ISDN

The Euro BRI voice switching daughtercard (VSB) is used to provide the European ISDN BRI voice port connections in Alcatel’s H.323 VoIP gateways. Because many aspects of this card are similar to the VSD T1/E1 previously discussed; see also VoIP Daughtercard Types on page 2-2 and Voice Switching Daughtercard — Digital on page 2-6.

Digital Signal Processors (DSPs) and Available Channels

Unlike the VSD T1/E1 VoIP daughtercard shown previously, the VSB card does not support any additional DIMMs, only the two standard DSP which provide the VSB with eight channels. See also Digital Signal Processors (DSPs), DIMMs and Available Channels on page 2-4 for a more details.

VSB Deadman Switch

There are two Deadman switches on the VSB daughtercards. One Deadman switch is for ports A and B (1 and 2) and the other is for ports C and D (3 and 4); on a VSX switching module with two VSBs, the relays switches on the second card would be for ports A and B (5 and 6) and for ports C and D (ports 7 and 8). For more details on the Deadman switch, see VSD Deadman Switch on page 2-8. Port numbers can vary depending on the VoIP switch configuration; see also VoIP Daughtercard Port Numbering Schemes on page 2-28.

♦ Note ♦

For the deadman switch to operate properly on the VSB, ports 1 and 3 must be configured as TE, and ports 2 and 4 must be configured as NT (ports are TE, NT, TE, NT).

VSB NT (LT)/TE Cross-Over Toggle Switch

There are four NT (LT) / TE Cross-Over toggle switches on the VSB daughtercards. Each switch is factory set to NT. For more information on ISDN terminators, seeVSB Jumpers on page 2-16, and also Chapter 3, “Network Dialing Schemes.”

The NT (LT)/TE toggle switch is physically similar to the VSD Cross-Over toggle switch on the VSB, but functionally different as it is used to select either a network terminator or terminal equipment as an endpoint. For a description of the Cross-Over toggle switch on the digital voice switching daughtercards, see VSD Cross-Over Toggle Switch on page 2-9, and Cabling on page 2-10.

NT (LT) means that each NT port emulates the “network” side or “line terminator” point of the ISDN connections to the ISDN/PSTN network, e.g., connections to PBX, Key Set, BRI TelSet, Group 4 (ISDN) facsimile machine. TE means that each TE port emulates the “terminal” side of the ISDN connections to the ISDN network, e.g., PBX, Key Set, CO (Central Office) switch and ISDN telephone switch.

VSB Pinouts

The Euro BRI VoIP daughtercard (VSB) uses the 8-pin, RJ-45 jacks in the figures and tables on the following page to connect the voice ports on the card to voice devices in the VoIP network that support digital connections, e.g., PBX, Key Set, BRI TelSets, Group 4 (ISDN) facsimile machine, CO (Central Office) switch, and ISDN telephone switch. For more details on the VSB pinouts, see VSD Pinouts on page 2-11.

Page 2-13

Voice Switching Daughtercard — Euro BRI ISDN

VSB Conﬁgured as TE

RJ-45 Speciﬁcations

Pin Number Standard Signal Name

1 Unused

2 Unused

3 Tx +

4 Rx +

5 Rx -

6 Tx -

7 Unused

VSB Conﬁgured as NT

RJ-45 Speciﬁcations

Pin Number Standard Signal Name

1 Unused

2 Unused

3 Rx +

4 Tx +

5 Tx -

6 Rx -

7 Unused

Page 2-14

Voice Switching Daughtercard — Euro BRI ISDN

VSB Front Panel

Each port has three corresponding LED indicators with link status displays as shown in the following illustration. VSBs, VSDs and VSAs cannot be installed in the same slot in an OSR. An MPX must also be installed in the OSR. Port numbers can vary depending on the VoIP switch configuration; see also VoIP Daughtercard Port Numbering Schemes on page 2-28.

All VSB daughtercards have three LED displays per voice port as follows:

FAIL: On when VSB fails or diagnostic test fails, or when VSB image download fails.

Off when VSB hardware is functional, or when VSB image download is OK.

VSB

B FAIL ERR LINK

ERR: On when VSB voice port link error occurs in line. This can be any bearer or Data channel type of error, e.g., out-offrame/loss of synchronization.

LINK: On when VSB voice port link to switch is connected.

Off when signal is lost VSB voice port link disconnected.

VSB (Voice Switching Digital) Euro BRI ISDN Daughtercard Front Panels

Page 2-15

Voice Switching Daughtercard — Euro BRI ISDN

VSB Jumpers

The VSB daughtercard requires jumpers for Network Terminator (NT), also referred to as Line Terminator (LT), and Terminal Equipment (TE), impedance, and power feeds, be set as follows and in the order presented, e.g., NT (LT)/TE jumper/switch settings must be set on the VSB before any other jumpers on the board. In general, the jumpers are set on a per port basis. Locations of the jumpers are illustrated below.

P31

Default

NT (LT) Setting

to Left

J18

VSB

J16

P46

(Jumper and toggle switch settings must match)

NT (LT)/ TE

Toggle Switches

J22

NT (LT)

Setting

to Right

J20

J25

J26

J27

J29

J28

P20

J30

Power Feeds

VSB Jumper Settings — NT (LT)/TE, Impedance and Power Feeds

The jumpers are factory set on the VSB daughtercard, and should be changed by customers only under the circumstances as listed below for each jumper. Note that, in general, only jumpers which can be set with shunts are identified and described. Jumpers should be set in the order in which they are presented.

♦ Note ♦

All VSB daughtercards are factory set to the defaults, including but not limited to jumpers J16 and J18, J25 through J28, as well as J29 and J30, and the corresponding NT (LT)/TE switches. This board is used only to provide Euro BRI ISDN (E1 ETSI) capabilities, and cannot be used in North America. See also VSD Jumpers on page 2-12.

Page 2-16

Voice Switching Daughtercard — Euro BRI ISDN

Network or Line Terminator (NT/LT) / Terminal Equipment (TE)

The NT/LT and TE jumpers (J29 and J30) are used to set the type of terminator on the VSB daughtercards. When the shunts are removed from these jumpers, NT (LT) is set as the terminator type; this is the default setting for both jumpers. When these jumpers and switches are set to either NT/LT or TE, the

voice port isdn protocol command must also be set to the same,

corresponding setting. For details on using this command, see Chapter 5, “VoIP Commands.”

♦ Note ♦

J29 and J30 must match settings on NT (LT)/TE (Cross-Over toggle switches. The default setting for the toggle and corresponding jumpers is NT (see illustration forVSB Jumpers on page 2-16.)

Default

(J29): NT (LT) on port 1; no shunts on pins 1_2.

(J29): NT (LT) on port 2; no shunts on pins 3_4.

(J30): NT (LT) on port 3; no shunts on pins 1_2.

(J30): NT (LT) on port 4; no shunts on pins 3_4.

TE (J29): TE on port 1; requires shunts on pins 1_2.

TE (J29): TE on port 2; requires shunts on pins 3_4.

TE (J30): TE on port 3; requires shunts on pins 1_2.

TE (J30): TE on port 4; requires shunts on pins 3_4.

For VSB jumpers J29 and J30, only the following five NT (LT)/TE port configurations are allowed:

• All four ports can be configured as TE (ports are TE, TE, TE, TE).

• All four ports can be configured as NT (Ports are NT, NT, NT, NT).

• Ports 1, 2, and 3 configured as TE and port 4 configured as NT (ports are TE, TE, TE, NT).

• Ports 1 and 3 configured as TE, and port 2 and 4 configured as NT (ports are TE, NT, TE, NT).

• Port 1 configured as TE, and port 2, 3, and 4 configured as NT (ports are TE, NT, NT, NT).

Impedance

The impedance jumpers (J16, J18, J20, and J22) are used to set the resistance to the alternating current on the VSB daughtercards. When the shunts are removed from these jumpers, the standard impedance of 100 Ohms will not be fed to the designated port; this is the default setting for all four jumpers.

Default

(J16): no impedance on port 1; no shunts on pins 1_2 and pins 3_4.

(J18): no impedance on port 2; no shunts on pins 1_2 and pins 3_4.

(J20): no impedance on port 3; no shunts on pins 1_2 and pins 3_4.

(J22): no impedance on port 4; no shunts on pins 1_2 and pins 3_4.

(J16): add impedance on port 1; requires shunts on pins 1_2 and pins 3_4.

(J18): add impedance on port 2; requires shunts on pins 1_2 and pins 3_4.

(J20): add impedance on port 3; requires shunts on pins 1_2 and pins 3_4.

(J22): add impedance on port 4; requires shunts on pins 1_2 and pins 3_4.

Page 2-17

Voice Switching Daughtercard — Euro BRI ISDN

Power Feeds

The power feed jumpers (J25, J26, J27 and J28) are used to set the power feeds on the VSB daughtercards. When the shunts are removed from these jumpers, no power is fed to the designated port; this is the default setting for all four jumpers.

Damage to the VSB daughtercards due to improper configuration of the power feeds is not covered by warranty.

For NT configurations only — The use of shunts to enable the power feeds on the VSB must be used with caution as they will substantially increase the voltage applied to the board, and may result in damage to the board and/or other components connected to the VSB. It is strongly recommended that Customer Support be contacted before installing shunts on jumpers J25 through J28.

♦ Caution ♦

Default

(J25): no power on port 1; no shunts on pins 1_2 and pins 3_4.

(J26): no power on port 2; no shunts on pins 1_2 and pins 3_4.

(J27): no power on port 3; no shunts on pins 1_2 and pins 3_4.

(J28): no power on port 4; no shunts on pins 1_2 and pins 3_4.

NT (J25): power feed on port 1; optional shunts on pins 1_2 and pins 3_4.

NT (J26): power feed on port 2; optional shunts on pins 1_2 and pins 3_4.

NT (J27): power feed on port 3; optional shunts on pins 1_2 and pins 3_4.

NT (J28): power feed on port 4; optional shunts on pins 1_2 and pins 3_4.

Other VSB Jumpers

The following jumpers are factory set on the VSB daughtercard and should not be changed by the customer unless under the direction of Customer Support.

Jumper No. Shunt Position Default Description

P20 Pins 1_2, Pins 4_5 yes Reserved

P31 Pins 2_3, Pins 4_5 yes Backplane interface (Hbus)

Page 2-18

P46 Pins 2_3 yes Reserved

Voice Switching Daughtercard — Analog

The Analog Voice Switching (VSA) daughtercard is used to provide the analog voice port connections in Alcatel’s H.323 VoIP gateways. Each VSA card includes either an FXS (Foreign Exchange Station) or FXO (Foreign Exchange Office) grand-daughtercard, or one of each depending on whether FXS signaling is needed to generate calls from POTS TelSets to a VoIP daughtercard, or FXO signaling is needed to generate calls to POTS TelSets from a VoIP daughtercard, or both. The FXS and FXO signaling protocols are used by the corresponding board types for which there are five VSA grand-daughtercard configurations:

• FXS

• FXS/FXS (Dual)

• FXO

• FXO/FXO (Dual)

• FXS/FXO (Mixed)

Depending on the VSA FXS and/or FXO grand-daughtercards used there can be either 2, 4, 6, or 8 ports per daughtercard. As shown below, in a VSX-FXS-FXS (dual) configuration, up to eight POTS TelSets can be connected, and in a VSX-FXO-FXO configuration up to four POTS PSTN lines can be connected. FXS and FXO grand-daughtercards can be used together (referred to as a mixed configuration) on one VSA card.

Specifically, the VSA port range for the OmniAccess 512 is 1 to 2 (single FXO), 1 to 4 (dual FXO/FXO or single FXS), 1 to 6 (mixed FXS/FXO), or 1 to 8 (dual FXS/FXS grand-daughtercards). The same applies to the OSR except the range can be from 1 to 16 based upon the grand-daughtercards installed (OSR full capacity installations not available this release). An MPX must also be installed in the OSR. In general, VSAs, VSDs and VSBs are not be mixed in the same VSX card in the same slot of an OSR; however, there are exceptions. See also the VSX Switching Module on page 2-25 and the VoIP Daughtercard Port Numbering Schemes on page 2-28.

All VSA daughtercards require 32 MB Flash memory, and for OSR configurations, 64 MB DRAM memory on the MPX. For power requirements, see VSX Switching Module on page 2-25.

The following certifications for the VSA daughtercards have been obtained to date: FCC Class A for OA-512 and OSR VSA-FXO.

Top View

Analog

Voice Ports

FXS module

DSP

Switch

Bus

Analog

Voice Ports

FXS module

DSP

Analog Voice Switching Daughtercard (VSA-FXS-FXS, Dual) — Top View

Page 2-19

Voice Switching Daughtercard — Analog

Analog voice switching daughtercards (VSAs) cannot be used without either installing an FXS or FXO granddaughtercard (or module).

FXO grand-daughtercards cannot be installed as ports 1 and 2 when used with an FXS grand-daughtercard.

FXS and FXO grand-daughtercards are not field upgradeable.

Calls between channels on the same VSA card use PCM (Pulse Code Modulation) instead of the H.323 protocol to process analog calls, and require two DSP channels to make the calls.

♦ Notes ♦

Analog

Voice Ports

Analog

Voice Ports

Analog

Voice Ports

FXO module

Top View

FXO module

DSP

Analog Voice Switching Daughtercard (VSA-FXO-FXO) — Top View

FXS module

DSP

Top View

Switch

Bus

Page 2-20

FXO module

Analog

Voice Ports

DSP

Analog Voice Switching Daughtercard (VSA-FXS-FXO, Mixed) — Top View

Switch

Bus

Voice Switching Daughtercard — Analog

VSA Front Panel

Each channel (port) has one corresponding LED indicator with link status displays as shown in the following illustration. Port numbers can vary depending on the VoIP switch configuration; see also VoIP Daughtercard Port Numbering Schemes on page 2-28.

All VSA boards have one link LED per voice port.

All VSA Voice Port LEDs display as follows:

Green On: VSA port link to switch is connected.

Green Off: VSA port link is disconnected.

Green Blinking: Phone connected to VSA port ringing; No Blinking: Phone connected to VSA port off hook.

For the FXO submodules, only two voice ports are active.

4 Ports / Single FXS module

VSA-4FXS

FXS

8 Ports / Dual FXS module

VSA-8FXS

FXS

2 Ports / Single FXO module

VSA-2FXO

FXO

Inactive

FXS

4 Ports / Dual FXO module

VSA-4FXO

FXO

6 Ports / Mixed FXS and FXO module

VSA

FXS

VSA — Front Panels

Inactive

FXO

Inactive

FXO

Inactive

Page 2-21

Voice Switching Daughtercard — Analog

VSA Pinouts

The following illustration shows the pinouts for the analog voice switching daughtercard (VSA) 8-pin, RJ-11 jacks used to connect the voice ports on the card to voice devices in the VoIP network that support analog connections, e.g., telephone and fax machine.

VSA TelSet RJ-11 Specifications

Pin Number Standard Signal Name

3 Ring

4 Tip

VSAs and Digital Signal Processors (DSPs), DIMMs and Available Channels

There are no DIMMs on the VSA, only DSPs on the FXS or FXO grand-daughtercards with a maximum of one channel per port. For more information on the DSPs, see Digital Signal Processors (DSPs), DIMMs and Available Channels on page 2-4 for more details.

VSAs and the Deadman Switch

Although VSA cards do not contain a Deadman switch, similar PSTN fallback call protection can be provided in the event of a power failure simply by plugging the analog VoIP switch into an Uninterruptable Power Supply (UPS). For a description of the Deadman switch on the digital voice switching daughtercards, see VSD Deadman Switch on page 2-8.

VSAs and Cross-Over Toggle Switches

There are no Cross-Over type toggle switches on the VSA daughtercards. For a description of the Cross-Over toggle switches on the digital voice switching daughtercards, see VSD Cross- Over Toggle Switch on page 2-9, and VSB NT (LT)/TE Cross-Over Toggle Switch on page 2-13.

Page 2-22

Voice Switching Daughtercard — Analog

VSA Jumpers

The VSA daughtercard requires the jumpers for Ringing Voltage and Ringing Frequency to be set as follows. For additional details on the VSA ringing voltage or ringing frequency, see also Chapter 5, “VoIP Commands.” In general, the jumpers are set on a per port basis. Locations of the jumpers are illustrated below.

♦ Notes ♦

All VSA daughtercards are factory set to the defaults for jumpers P33 and P34. The defaults are applicable to US and Europe versions of the board. See also VSD Jumpers on page 2-12.

Default jumper settings on VSA daughtercards can be used in Europe, but ringing voltage and frequency must

also match requirements as per specification of the equipment to be used with a VSA, e.g., telephone,

facsimile machine, terminal equipment (TE), PBX, etc.

VSA

P31

MADE IN USA

P46

P33

P34

Ringing Voltage

Ringing

Frequency

P20

VSA Jumper Settings — Ringing Voltage and Ringing Frequency

Ringing Voltage

The jumper for ringing voltage (P33) is used to set the continental ring tone in Vrms (voltage mean root square) on the VSA daughtercards.

Default

: 75 Vrms; requires shunts on pins 2_3 and pins 5_6 (No. Amer./European Spec.).

45 Vrms; requires shunts on pins 2_3 and pins 4_5 (European Specification).

86 Vrms; requires shunts on pins 1_2 and pins 5_6 (European Specification).

Page 2-23

Voice Switching Daughtercard — Analog

Ringing Frequency

The jumper for ringing frequency (P34) is used to set the frequency of the continental ring tone in Hertz (Hz) on the VSA daughtercards.

Default

: 20 Hz; requires shunts on pins 2_3 and pins 5_6 (No. Amer./European Spec.).

16 Hz; requires shunts on pins 2_3 and pins 4_5 (European Specification).

25 Hz; requires shunts on pins 1_2 and pins 5_6 (European Specification).

Other VSA Jumpers

The following jumpers are factory set on the VSA daughtercard and should not be changed by the customer unless under the direction of Customer Support.

Jumper No. Shunt Position Default Description

P20 Pins 1_2, Pins 4_5 yes Reserved

P31 Pins 2_3, Pins 4_5 yes Backplane interface (Hbus)

P46 Pins 2_3 yes Reserved

Page 2-24

VSX Switching Module

As illustrated on the next two pages, the VSX switching module accepts up to two VoIP daughtercards, and can only be installed in the Omni Switch/Router as follows:

• A maximum of two (2) VoIP daughtercards can be installed per OSR switch.

• One VoIP daughtercard can be installed per VSX in a single slot.

• Two (2) VoIP daughtercards can be installed in

S3/1 (left) and S3/2 (right) a VSX in a

single slot.

With two digital daughtercards in a VSX in an OSR, for instance, it is possible to process up to 96 calls (T1) or 120 calls (E1) per VSX switching module. For port scalability, up to seven VSX switching modules with two VoIP daughtercards each of the same type can be installed per OSR (maximum capacity configuration not available this release). Pertinent specifications for the VSX switching module are as follows.

♦ Notes ♦

VoIP daughtercards cannot be installed in an HSX for interfacing with an OSR; however, two VoIP daughtercards (e.g., two VSBs) can be installed in a VSX (HSXH) into an OSR with an MPX card, if necessary.

In OSR configurations, the use of an HRE-X device may be required. For more information, see the Omni Switch/Router user manual.

VSX Technical Speciﬁcations

Data Rates Supported 1024 and 2048 Kbps

Clocking Internal and External

MAC Addresses Supported 4096

H.323 IP Addresses Supported 2

Connections Supported ISDN, QSIG master/slave, ETSI, FXS/FXO

Maximum Number of Simultaneous Voice/Data/Fax Channels Supported

Cable Supported RJ-48C and Telset RJ-11

Power Consumption 5.50 amps

120

Additional 2.0 amps per VSD or VSB

Additional 3.55 amps per VSA daughtercard

Additional 5.25 amps per VSA (with one FXS or FXO grand-daughtercard)

Additional 7.0 amps per VSA (with two FXS or FXO grand-daughtercards)

Page 2-25

VSX Switching Module

VoIP Daughtercard Port Numbering Schemes

The following table is a representation of the port numbering schemes for all VoIP daughtercard configurations in either an OmniAccess 512 or an Omni Switch/Router. Although FXS and FXO installations must include a VSA daughtercard, for purposes of this table only, the VSA card is not specified; a maximum of two FXS or two FXO grand-daughtercards can be installed per VSA daughtercard. When determining valid port numbers, use this list as a reference:

• Each VSD has two ports.

• Each VSB has four ports.

• Each VSA FXS has four ports.

• Each VSA FXO has two ports.

OSR configurations are currently limited to a maximum capacity of two daughtercards per switch. Configuration options included one VXS module with two VoIP daughtercards, or two VXS modules and one VoIP daughtercard per VSX. Also, not all configurations shown below may be currently available for purchase.

♦ Notes ♦

See also VSD Front Panel on page 2-7, VSB Front Panel on page 2-15, VSA Front Panel on page 2-21; these drawings illustrate how VoIP daughtercard ports are numbered. For information on configuring the ports using CLI commands, see Chapter 5, “VoIP Commands”.

Switch VSD VSB FXS FXO Valid Port/Configuration Numbers

OA-512 1 - - - A-B (Note: Port numbers must be conﬁgured as 1 and 2; A=1

and B=2.)

OA-512 - 1 - - A, B, C, D (Note: Port numbers must be conﬁgured as 1, 2, 3

and 4; A=1, B=2, C=3, D=4.)

OA-512 - - 1 - 1, 2, 3, 4 (FXS)

OA-512 - - 2 - 1-4 (FXS), 5-8 (FXS)

OA-512 - - - 1 1, 2 (FXO)

OA-512 - - - 2 1-2 (FXO), 3-4 (FXO)

OA-512 - - 1 1 1-4 (FXS), 5-6 (FXO)

OSR 1 - - - A-B (Note: Port numbers must be conﬁgured as 1 and 2; A=1

and B=2.)

OSR 2 - - - A-B (S3/1 position), A-B (S3/2 position);

(Note: Port numbers in S3/1 must be conﬁgured as 1 and 2; A=1, B=2. In S3/2 port numbers must be conﬁgured as 3 and 4; A=3, B=4.)

OSR - 1 - - A, B, C, D (Note: Port numbers must be conﬁgured as 1, 2, 3

and 4; A=1, B=2, C=3, D=4.)

Page 2-28

VoIP Daughtercard Port Numbering Schemes

Switch VSD VSB FXS FXO Valid Port/Configuration Numbers

OSR - 2 - - A-D (S3/1 position), A-D (S3/2 position);

(Note: Port numbers in S3/1 must be conﬁgured as 1, 2, 3, and 4; A=1, B=2, C=3, D=4. In S3/2 port numbers must be conﬁgured as 5, 6, 7, and 8; A=5, B=6, C=7, D=8.)

OSR - - 1 - 1-4 (FXS), (S3/1)

OSR - - 2 - 1-4 (FXS), (S3/1), 5-8 (S3/2)

OSR - - 3 - 1-12 (FXS-FXS-FXS)

OSR - - 4 - 1-16 (FXS-FXS-FXS-FXS)

OSR - - - 1 1-2 (FXO), (S3/1)

OSR - - - 2 1-2 (FXO), 3-4 (FXO), (S3/1)

OSR - - - 3 1-2 (FXO), 3-4 (FXO), (S3/1), 5-6 (S3/2)

OSR - - - 4 1-8 (FXO-FXO-FXO-FXO)

OSR - - 1 1 1-4 (FXS), 5-6 (FXO)

OSR - - 1 2 1-4 (FXS), 5-6 (FXO), 7-8 (FXO)

OSR - - 1 3 1-4 (FXS), 5-6 (FXO), 7-8 (FXO), 9-10 (FXO)

OSR - - 2 1 1-4 (FXS), 5-8 (FXS), 9-10 (FXO)

OSR - - 2 2 1-4, 5-6 (FXS-FXO), 7-10, 11-12 (FXS-FXO)

or 1-8 (FXS-FXS), 9-12 (FXO-FXO)

OSR - - 3 1 1-4 (FXS), 5-6 (FXO), 7-14 (FXS)

or 1-12 (FXS-FXS-FXS), 13-14 (FXO)

Page 2-29

VoIP Daughtercard Port Numbering Schemes

Page 2-30

3 Network Dialing

Schemes

Introduction

This chapter contains information on selecting and configuring Alcatel VoIP Network Dialing Schemes (AVNDS) which are used to translate dialed digits into IP addresses on the switch. At least one dialing scheme must be configured to support a Voice over IP network.

The dialing scheme examples discussed in this chapter are daughtercard centric, and typically consist of two PBXs with corresponding voice switching daughtercards each connected by one incoming only and one outgoing only trunk. In most cases the PBX is assumed to be trunked to the North American PSTN (Public Switched Telephone Network); however, some examples have voice daughtercards connected to the PSTN. It should be presumed also that all calls going to the PSTN are directed by Telco Central Offices. The WAN links between the switches (or some other device) are via the WSM or WSX modules which provide the ports, e.g., T1, E1, for data communications. The voice daughtercards (VSD, VSB and VSA) provide the telephony ports, e.g., T1, E1, Euro ISDN BRI, FXO and FXS for voice communications. Except where specified otherwise, it should be assumed that the VoIP daughtercards used in the examples are VSDs.

Variations to the dialing scheme configurations entail other likely scenarios in a VoIP network, including the use of hunt groups, site prefixes, strip digits, fax over IP and caller ID. Dialing schemes for special configurations, such as using VoIP in the switch with the OmniPCX 4400, are provided as well. All dialing schemes can be used in OmniAccess 512 and Omni Switch/Router configurations.

To simplify the configuration process, a VSM (Voice Switching Module) partial text-based ASCII configuration boot file ( partial boot file contains the specific CLI commands needed to implement a selected dialing scheme, and should be merged with the complete master boot file (vsmboot_master.asc), modified accordingly and then installed on the switch. Refer to Chapter 4, “Setup and Installation,” for further details and an example boot file configuration. For specific details on the VoIP text-based command line interface (CLI) commands relative to the boot files and dialing schemes, see Chapter 5, “VoIP Commands”.

Companies using Alcatel’s VoIP feature are responsible for programming and testing all dialing schemes to reduce the likelihood or to eliminate the possibility of toll fraud from the PSTN and Emergency 911 processing.

vsmboot.asc) has been created for each dialing scheme. Each

♦ Caution ♦

Page 3-1

Introduction

The AVNDS consist of the following CLI command groups:

• H.323 endpoint destinations

— these commands describe the IP address of every

H.323 device in the H.323 VoIP network.

• H.323 local channel destinations

— these commands describe every available channel

on a voice daughtercard port in which to send calls.

• Phone Group

— these commands describe every phone/fax number allowed on the

H.323 VoIP network.

• Numbering Plan

— these commands relate phone groups to destinations (H.323

endpoint and local channel destinations).

The AVNDS must include all CLI commands (approx. 20 commands) concerning voice destinations, phone groups and numbering plans to render a dialing scheme operational; however, some AVNDS will include other “non-AVNDS” commands in the matching

vsmboot.asc files,

as is the case with dialing scheme examples 12 and 17. For more details, see also Chapter 4, “Setup and Installation,” and Chapter 5, “VoIP Commands.”

Page 3-2

How to Select a Network Dialing Scheme (AVNDS)

The tables below contain a list of the dialing scheme examples; use the decision criteria in the far right column of each table to determine the most appropriate dialing scheme to follow when configuring the network for VoIP in the switch. The dialing scheme examples, of which there are 26, are discussed in this chapter in numerical order, but are categorized into three distinct types:

• VoIP Networks without PSTN (Dialing Schemes 1-12)

Dialing scheme examples in this group do not connect to the PSTN. It is assumed that the PBX handles the routing of the call to the VoIP network. The first two examples are considered basic dialing schemes, while the remaining examples in this group demonstrate more complex VoIP dialing scheme concepts, such as how to use hunt groups (to multiply and split T1 lines), strip digits, or an H.323 gatekeeper.

• VoIP Networks with PSTN (Dialing Schemes 13-18)

Dialing scheme examples in this group connect the voice daughtercards to the North American PSTN, and cover the use of strip digits, fax over IP, and caller ID (forwarding and static). International (ISDN) PSTN and Caller ID Forwarding not available this release.

• VoIP Networks with Interoperability (Dialing Schemes 19-26)

Dialing scheme examples in this group allow VoIP networks to work with other functionally related equipment including H.323 gateways, H.323 endpoints, the OmniPCX 4400 and assorted PBXs.

Introduction

No. Dialing Scheme Examples / VoIP Networks without PSTN Decision Criteria

1 Four Digit Extensions and Two Voice Daughtercards Basic VoIP Network

2 Four Digit Extensions and Three Voice Daughtercards Expanded VoIP Network

3 Hunt Groups — One Hunt Group (48 channels across two T1s) One Hunt Group Per T1 Voice

Daughtercard

4 Hunt Groups — One Hunt Group (60 channels across two E1s) One Hunt Group Per E1 Voice

Daughtercard

5 Hunt Groups — One Hunt Group (96 channels across four T1s) One Hunt Group Across Two Voice

Daughtercards

6 Hunt Groups — One Hunt Group (144 channels across six T1s) One Hunt Group Across Three

Voice Daughtercards

7 Hunt Groups — Four Hunt Groups (12 channels per group) Fractional (1/2) T1 Hunt Groups

8 Hunt Groups — 48 Individual Hunt Groups (One channel per group) Fractional (individual channel) T1

Hunt Groups

9 Strip Digits — Trunk Groups and Mixed Length Extensions Unique mixed length extensions

10 Strip Digits — Trunk Groups and Two Strip Digits Common extensions,

Unique, two digit site preﬁx

11 Strip Digits — Trunk Groups and Eleven Digit Extensions (NANP-like) Common extensions

Eleven digit local extensions to simulate NANP dialing

12 H.323 Gatekeeper Unique extensions

Complex VoIP Network with H.323 gatekeeper

Page 3-3

Introduction

No. Dialing Scheme Examples / VoIP Networks with PSTN Decision Criteria

13 North American PSTN — Four Digit Extensions and Direct Inward Dial

(DID)

14 International (ISDN) PSTN — Four Digit Extensions and Direct Inward

Dial (DID).

15 North American PSTN — Eleven Digit Extensions Eleven Digit NANP Extensions

16 North American PSTN — Fax over IP Network Toll-Saving Fax Calls

17 North American PSTN — VSD/VSA Mixed Mixed Digital and Analog Voice

18 North American PSTN — Caller ID (Static). Analog Voice Daughtercard gen-

Unique NANP extensions Unique Site Preﬁx DID

No. Amer and Intl. Sites DID

with PSTN

Daughtercards. PSTN at remote site.

erating predetermined static caller ID

No. Dialing Scheme Examples / VoIP Networks with Interoperability Decision Criteria

19 H.323 Gateway — Microsoft NetMeeting (w/o FastStart) VoIP to 3rd Party H.323 Soft-

ware

20 H.323 Gateway — Cisco Routers VoIP to 3rd Party H.323 Hard-

ware

21 H.323 Gateway — OmniPCX 4400 VoIP to OmniPCX LIOE card

22 Omni PCX 4400 — E1 QSIG Interoperating via E1 QSIG

23 Omni PCX 4400 — Euro PRI Interoperating via Euro PRI

24 Other PBXs — T1 Interoperating with 3rd Party

PBX via T1.

25 Other PBXs — Euro BRI Interoperating with 3rd Party

PBX via Euro BRI (E1 ETSI)

26 Other PBXs — European VSD/VSA Mixed Interoperating with European

Digital and Analog Voice Daughtercards.

All dialing schemes in this chapter can be modified to be used with the VSD, VSB and VSA voice switching daughtercards with the following exceptions:

• Dialing schemes No. 7 and 8 (Fractional T1 Hunt Groups) apply only to VSD and VSB daughtercards. Fractional type hunt groups do not apply to VSAs because analog channels can only be combined, not multiplied or split.

• Dialing schemes No. 17 (Mixed Digital and Analog) and No. 18 (Static Caller ID) apply to North American configurations using VSA daughtercards. Dialing scheme No. 26 (Mixed Digital and Analog) applies to a European VSA configuration.

• Dialing scheme No. 25 applies only to VSB daughtercards (European only).

• Except for when local channel destinations are used, all AVNDS commands function with the H.323 endpoints, e.g., OmniPCX, Cisco Routers, Microsoft NetMeeting.

Page 3-4

Network Dialing Scheme VoIP Features

The table below lists dialing plans that use particular VoIP features. (See also AVNDS Master List of Features by CLI Command on page 3-67.) Descriptions of the dialing schemes in this

chapter are intended to serve as guidelines in the development of enterprise-specific network VoIP dialing schemes.

VoIP Feature Dialing Scheme

H.323 gateway to voice daughtercard (A) All

H.323 gateway to H.323 gatekeeper (RADVision) (B) 12

H.323 gateway to H.323 device (C) 19, 21

Local channel — 48 individual hunt groups (One channel per group) (D) 8, 17, 18

Local channel — four hunt groups (12 T1 channels per group) (E) 7

Local channel — two hunt groups (24 T1 channels per group) (F) 1-3, 5, 7, 9-16, 19-24

Local channel — one hunt group (48 channels across two T1s) (G) 5, 6

Local channel — one hunt group (60 channels across two E1s) (H) 4

Site preﬁx — no site preﬁx (I) 1-9, 12, 14, 15, 19-24

Site preﬁx — single or multiple digits (J) 9-11, 13-19, 21

Voice phone group type — three digit local extensions (K) 9, 10

Voice phone group type — four digit local extensions (L) 1-8, 11, 12, 20

Voice phone group type — eleven digit local extensions (M) 11, 19, 21-24

Voice phone group type — NANP extensions (N) 13-18

Voice phone group type — INTL extension (O) 14, 22, 23

Voice phone group type — PSTN NANP (P) 13-18

Voice phone group type — PSTN International (INTL) (Q) 25

Strip digit length — no strip digits (R) 1-9, 12, 17-24

Strip digit length — 2 (T) 10

Strip digit length — 7 (V) 11, 13-16, 19-24

Digital Interface type — T1 (W) 1-3, 5-16, 19-24

Digital Interface type — E1 (QSIG) (X) 22

Digital Interface type — E1 ISDN PRI (Euro PRI) (Y) 4, 14, 23

Digital Interface type — Euro BRI (Z)25

Digital Interface type — FXS (AA) 17, 18

Digital Interface type — FXO (AB) 17, 18

Introduction

Page 3-5

Introduction

Trunk List for AVNDS Examples

The trunk list below is also provided as a general reference guideline to each of the diagrams used in the dialing scheme examples. Note that most examples use only one or two PBXs, and one or two voice daughtercards.

Lines (A): Telephone/Fax lines off of PBX #1.

Lines (B): Telephone/Fax lines off of PBX #2.

Lines (C): Telephone/Fax lines off of PBX #3.

Trunk (D): Inbound trunk to PBX #1.

Trunk (E): Inbound trunk to PBX #2.

Trunk (F): Inbound trunk to PBX #3.

Trunk (G): Inbound trunk to Voice Daughtercard #1.

Trunk (H): Inbound trunk to Voice Daughtercard #2.

Trunk (I): Inbound trunk to Voice Daughtercard #3.

Trunk (X): Inbound trunk to Voice Daughtercard #4.

Trunks (J), (K): Inbound/Outbound trunk (to Voice Daughtercard #1).

Trunks (L), (M): Inbound/Outbound trunk (to Voice Daughtercard #2).

Trunks (N), (O): Inbound/Outbound trunk (to Voice Daughtercard #3).

Trunks (P), (Q): Inbound/Outbound trunk (to Voice Daughtercard #4).

Trunk (R), (S): Inbound/Outbound trunk (between Voice Daughtercard #1 and PSTN)

Trunk (T), (U): Inbound/Outbound trunk (between Voice Daughtercard #2 and PSTN)

Trunk (V): Inbound/Outbound trunk (between Voice Daughtercard #3 and PSTN)

Trunk (W): Inbound/Outbound trunk (between Voice Daughtercard #4 and PSTN)

Ethernet WAN (Z): Inbound Ethernet (WAN) to H.323 Device

Page 3-6

VoIP Networks without PSTN — Example 1

Four Digit Extensions and Two Voice Daughtercards

This is one of the simplest dialing schemes to implement in a VoIP network. It uses two voice switching daughtercards to translate four digit extensions. Extensions are unique across the entire enterprise network, and the PBX handles all calls to the PSTN. Since incoming and outgoing trunks are separated, this dialing scheme guarantees that no inseize collisions will occur.

Extensions

off of PBX #1

10001999

(A)

PBX #1

1000 to 1999

T1 T1

(D)

(G)

Voice

Daughtercard

IP 192.168.11.2

Port 1720

WAN

PBX #2

Extensions

off of PBX #2

20002999

(B)

2000 to 2999

(E)

(H)

Voice

Daughtercard

IP 192.168.12.2

Port 1720

Example 1 — Four Digit Extensions and Two Voice Daughtercards

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on trunk (D) and then uses these digits to route calls to lines (A).

— Routes calls starting with 1 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 2 to trunk (G), and then the VoIP network uses these digits to route calls to trunk (E).

— Expects to receive four digits on trunk (E) and then uses these digits to route calls to lines (B).

— Routes calls starting with 1 to trunk (H), and then the VoIP network uses these digits to route calls to trunk (D).

Page 3-7

VoIP Networks without PSTN — Example 1

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — two hunt groups (24 channels per group/T1) (F)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — T1 (W)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-8

VoIP Networks without PSTN — Example 2

Trunk Groups and Three Voice Daughtercards

This dialing scheme is used to set up additional sites on an existing VoIP network. Individual calls are routed to more than one possible destination, e.g., from Trunk G to Trunk E or Trunk F, depending on the number dialed. All telephone numbers in this example are unique across the VoIP network.

Extensions

off of PBX #1

1000-

1999

Extensions

off of PBX #2

20002999

Extensions

off of PBX #3

30003999

(A)

(B)

(C)

PBX #1

PBX #2

PBX #3

1000-1999

2000-2999

3000-3999

(D)

(G)

(E)

(H)

(F)

(I)

Voice

Daughtercard

IP 192.168.11.2

Port 1720

Voice

Daughtercard

IP 192.168.12.2

Port 1720

Voice

Daughtercard

IP 192.168.13.2

Port 1720

WAN

Example 2 — Trunk Groups and Three Voice Daughtercards

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration PBX #3 Conﬁguration

— Expects to receive four digits on trunk (D) and then uses these digits to route calls to lines (A).

— Routes calls starting with 1 to lines (A).

— Routes calls starting with 2 or 3 to trunk (G), and then the VoIP network uses these digits to route calls to trunks (E) or (F), respectively.

— Expects to receive four digits on trunk (E) and then uses these digits to route calls to lines (B).

— Routes calls starting with 2 to lines (B).

— Routes calls starting with 1 or 3 to trunk (H), and then the VoIP network uses these digits to route calls to trunks (D) or (F), respectively.

— Expects to receive four digits on trunk (F) and then uses these digits to route calls to lines (C).

— Routes calls starting with 3 to lines (C).

— Routes calls starting with 1 or 2 to trunk (I), and then the VoIP network uses these digits to route calls to trunks (D) or (E), respectively.

Page 3-9

VoIP Networks without PSTN — Example 2

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — two hunt groups (24 channels per group/T1) (F)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — T1 (W)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-10

VoIP Networks without PSTN — Example 3

One Hunt Group (48 Channels Across Two T1s)

This dialing scheme uses one hunt group spanning two T1 lines to make a single 48 channel trunk.

Hunt groups are called voice numbering plans in the AVNDS. Voice numbering plans relate phone groups and destinations. In this example, the voice daughtercards are using the top down hunt method to determine where new calls will be routed.

Extensions

off of PBX #1

10001999

Extensions

off of PBX #2

20002999

(A)

(B)

PBX #1

PBX #2

1000 to 1999

Hunt Group

2000 to 2999

(J)

(K)

(L)

(M)

Daughtercard

IP 192.168.11.2

Daughtercard

IP 192.168.12.2

Example 3 — One Hunt Group (48 Channels Across Two T1s)

Voice

Port 1720

WAN

Voice

Port 1720

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on trunks (J) or (K) and then uses these digits to route calls to lines (A).

— Routes calls starting with 1 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 2 to trunks (J) or (K), and then the VoIP network uses these digits to route calls to trunks (L) or (M) according to the hunt method used

— Expects to receive four digits on trunks (L) or (M) then uses these digits to route calls to lines (B).

— Routes calls starting with 1 to trunks (L) or (M), and then the VoIP network uses these digits to route calls to trunks (J) or (K) according to the hunt method used.

Page 3-11

VoIP Networks without PSTN — Example 3

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — two hunt groups (24 channels per group/T1) (F)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — T1 (W)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-12

VoIP Networks without PSTN — Example 4

One Hunt Group (60 Channels Across Two E1s)

This dialing scheme uses one hunt group spanning two E1 (Euro PRI) trunks to make a single 60 channel trunk.

Extensions

off of PBX #1

10001999

Extensions

off of PBX #2

20002999

(A)

(B)

PBX #1

PBX #2

1000 to 1999

Hunt Group

2000 to 2999

Euro

PRI

Euro

PRI

Euro

PRI

Euro

PRI

(J)

(K)

IP 192.168.11.2

(L)

(M)

IP 192.168.12.2

Daughtercard

Example 4 — One Hunt Group (60 Channels Across Two E1s)

LEGEND for Diagram Components

Voice

Port 1720

WAN

Voice

Port 1720

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on trunks (J) or (K) and then uses these digits to route calls to lines (A).

— Routes calls starting with 1 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 2 to trunks (J) or (K), and then the VoIP network uses these digits to route calls to trunks (L) or (M) according to the hunt method used

— Expects to receive four digits on trunks (L) or (M) then uses these digits to route calls to lines (B).

— Routes calls starting with 1 to trunks (L) or (M), and then the VoIP network uses these digits to route calls to trunks (J) or (K).according to the hunt method used.

Page 3-13

VoIP Networks without PSTN — Example 4

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — one hunt group (60 channels across two E1s) (H)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — E1 ISDN PRI (Euro PRI) (Y)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

voice signaling companding (page 5-77)

Page 3-14

VoIP Networks without PSTN — Example 5

One Hunt Group (96 Channels Across Four T1s)

In this dialing scheme, one hunt group spans four T1 lines using two voice switching daughtercards (spanning two T1 lines each). In this example, the cards are installed in a single VSX motherboard in the same Omni Switch/Router to provide four T1 lines connected to a PBX.

Voice

(L)

Daughtercard

IP 192.168.11.2

Port 1720

Voice

Daughtercard

IP 192.168.12.2

Port 1720

Extensions

off of PBX #1

10001999

(A)

PBX #1

1000 to 1999

Hunt Group

(J)

(K)

(M)

Extensions

off of PBX #2

20002999

(B)

2000 to 2999

(E)

(I)

Voice

Daughtercard

IP 192.168.13.2

Port 1720

Example 5 — One Hunt Group (96 Channels Across Four T1s)

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

PBX #2

— Expects to receive four digits on trunks (J), (K), (L), or (M), and then uses these digits to route calls

to lines (A).

— Routes calls starting with 1 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 2 to trunks (J), (K), (L), or (M), and then the VoIP network uses these digits to route calls to trunk (E).

— Expects to receive four digits on (E) and then uses these digits to route calls to lines (B).

— Routes calls starting with 1 to trunk (I), and then the VoIP network uses these digits to route calls to trunks (J), (K), (L), or (M) according to the hunt method used.

WAN

Page 3-15

VoIP Networks without PSTN — Example 5

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — two hunt groups (24 channels per group/T1) (F)

Local channel — one hunt group (48 channels across two T1s) (G)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — Euro BRI (Z)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-16

VoIP Networks without PSTN — Example 6

One Hunt Group (144 Channels Across Six T1s)

This dialing scheme has six T1 lines connected to one PBX. In this dialing scheme, one hunt group spans six T1 lines using three voice switching daughtercards (spanning two T1 lines each). In this example, the cards are installed in a two VSX motherboards in the same Omni Switch/Router to provide six T1 lines connected to a PBX.

Hunt groups are called voice numbering plans in the AVNDS. Voice numbering plans relate phone groups and destinations. In this example, the voice daughtercards are using the round robin hunt method to determine where new calls will be routed.

Extensions

off of PBX #1

10001999

Extensions

off of PBX #2

20002999

(A)

(B)

PBX #1

PBX #2

1000 to 1999

Hunt Group

2000 to 2999

T1 T1

(J)

(K)

(L)

(M)

(N)

(O)

(E)

(X)

Voice

Daughtercard

IP 192.168.11.2

Port 1720

Voice

Daughtercard

IP 192.168.13.2

Port 1720

Voice

Daughtercard

IP 192.168.14.2

Port 1720

Voice

Daughtercard

IP 192.168.12.2

Port 1720

WAN

Example 6 — One Hunt Group (144 Channels Across Six T1s)

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on trunks (J), (K), (L), (M), (N) or (O), and then uses these digits to

route calls to lines (A).

— Routes calls starting with 1 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 2 to trunks (J), (K), (L), (M), (N) or (O), and then the VoIP network uses these digits to route calls to trunk (E).

— Expects to receive four digits on trunk (E) and then uses these digits to route calls to lines (B).

— Routes calls starting with 1 to trunks (X), and then the VoIP network uses these digits to route calls to trunks (J), (K), (L), (M), (N) or (O) according to the hunt method used.

Page 3-17

VoIP Networks without PSTN — Example 6

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — one hunt group (48 channels across two T1s) (G)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — T1 (W)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-18

VoIP Networks without PSTN — Example 7

Four Hunt Groups (12 Channels Per Hunt Group)

This dialing scheme is used to split a T1 line in half, and demonstrates one way of having redundant T1 lines on one switch. In this example, one hunt group is half of a T1 line, or 12 channels. Each T1 trunk is split into an incoming and outgoing hunt group.

Dialing Scheme Examples 7 and 8 (showing fractional T1 hunt groups) apply only to digital (VSD) and Euro BRI (VSB) voice switching daughtercards. See Chapter 2, “VoIP Daughtercards” for a description of the various daughtercards, and Chapter 4, “Setup and Installation” for details on installation.

Extensions

off of PBX #1

10001999

Extensions

off of PBX #2

20002999

(A)

PBX #1

1000 to 1999

4 Hunt Groups

Channels

1...12

13...24

1...12

13...24

(D)

Daughtercard

IP 192.168.11.2

(G)

Port 1720

4 Hunt Groups

Channels

1...12

13...24

1...12

13...24

(E)

Daughtercard

IP 192.168.12.2

Port 1720

(B)

PBX #2

2000 to 2999

(H)

Example 7— Four Hunt Groups (12 Channels Per Hunt Group)

Voice

WAN

Voice

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on channels 1 through 12 on trunks (D) or (G), and then uses these digits to route calls to lines (A).

— Routes calls starting with 1 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 2 to channels 13 through 24 on trunks (D) or (G), and then the VoIP network uses these digits to route calls to trunks (E) or (H).

— Expects to receive four digits on channels 1 through 12 on trunks (R) or (H), and then uses these digits to route calls to lines (B).

— Routes calls starting with 1 to channels 13 through 24 on trunks (E) or (H), and then the VoIP network uses these digits to route calls to trunks (D) or (G).

Page 3-19

VoIP Networks without PSTN — Example 7

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — four hunt groups (12 channels per group/T1) (E)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — T1 (W)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-20

VoIP Networks without PSTN — Example 8

48 Individual Hunt Groups (One Channel Per Group)

This dialing scheme shows how to divide a single T1 line into smaller (or fractional T1) trunk groups. Additionally, each channel has a unique telephone number and is also associated with a single telephone number and a single channel. Often this dialing scheme is used to test individual channels on a T1 line, but it can also be used to bypass hunt group behavior. Since each hunt group has only one channel, hunting is, in effect, disabled.

Extensions

off of PBX #1

10011024

Extensions

off of PBX #2

20012024

(A)

(B)

PBX #1

PBX #2

Ext. #

(G)

1001 1002

. . .

1024

(D)

48 Hunt Groups

Ext. #

(H)

2001

2002

. . .

2024

(E)

Channel

1 2

. . .

1 2

. . .

Channel

1 2

. . .

1 2

. . .

Voice

Daughtercard

IP 192.168.11.2

Port 1720

WAN

Voice

Daughtercard

IP 192.168.12.2

Port 1720

Example 8 — 48 Individual Hunt Groups (One Channel Per Group)

Page 3-21

VoIP Networks without PSTN — Example 8

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on channels 1 through 24 on trunk (D), and then uses these digits to route calls to lines (A.

— Routes calls starting with 1 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 2 to channels 1 through 24 on trunk (G), and then the VoIP network uses these digits to route calls to trunk (E).

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — individual hunt groups (48 channels per group/T1) (D)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — T1 (W)

LEGEND for Diagram Components

— Expects to receive four digits on channels 1 through 24 on trunk (E), and then uses these digits to route calls to lines (B).

— Routes calls starting with 1 to channels 1 through 24 on trunk (H), and then the VoIP network uses these digits to route calls to trunk (D).

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-22

VoIP Networks without PSTN — Example 9

Trunk Groups and Mixed Length Extensions

This is another dialing scheme that is relatively simple to implement in a VoIP network, and demonstrates how to mix different extension lengths in one dialing scheme. It uses two voice switching daughtercards to translate a single digit trunk prefix and three digit extensions. The single digit site prefix, rather than the three digits extensions, are unique across the VoIP network. The site prefix digit is used to send the VoIP calls to the correct PBX node. When a caller dials a site prefix, e.g., 1, it routes the call to the corresponding PBX and then dials a prefix to get a specific trunk.

Extensions

off of PBX #1

13001899

Extensions

off of PBX #2

300899

PBX #1

(A)

not stripped

“1”+ (300 to 899)

(D)

(G)

IP 192.168.11.2

PBX #2

(B)

300 to 899

(E)

(H)

IP 192.168.12.2

Example 9 — Trunk Groups and Mixed Length Extensions

Voice

Daughtercard

Port 1720

WAN

Voice

Daughtercard

Port 1720

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on trunk (D), and then uses these digits to calls to line (A).

— Routes calls starting with 1 to lines (A). — Routes calls starting with 3 to lines (B).

— Routes calls starting with 3 to trunk (G), and then the VoIP network uses these digits to route calls to trunk (E).

— Expects to receive three digits on trunk (E), and then uses these digits to route calls to line (B).

— Routes calls starting with 1 to trunk (H), and then the VoIP network uses these digits to route calls to trunk (D).

Page 3-23

VoIP Networks without PSTN — Example 9

Remarks

In the CLI commands, trunk groups are referred to as Site Prefix. Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — two hunt groups (24 channels per group/T1) (F)

Site preﬁx — no site preﬁx (I)

Site preﬁx — single or multiple digits (J)

Voice phone group type — three digit local extensions (K)

Strip digit length — no strip digits (R)

Digital Interface type — T1 (W)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group site prefix (page 5-238)

voice phone group site prefix digits (page 5-239) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-24

VoIP Networks without PSTN — Example 10

Strip Digit Length (2)

In this dialing scheme, the PBX uses the first two digits received to route calls. The PBX first dials an “8” to go to the VoIP network. The 2nd digit dialed (“1” or “2”) determines the site (PBX) to which the call is sent. A “1” means the call goes to PBX# 1, and a “2” means the call goes to PBX #2. The two-digit prefix is stripped before the destination voice switching daughtercard sends the digits to the PBX.

stripped

digits

(81) + (000 to 799)

(D)

(G)

Voice

Daughtercard

IP 192.168.11.2

Port 1720

WAN

Extensions

off of PBX #1

000799

PBX #1

(A)

Extensions

off of PBX #2

000799

(B)

(82) + (000 to 799)

stripped

digits

(H)

(E)

Voice

Daughtercard

IP 192.168.12.2

Port 1720

Example 10 — Strip Digit Length (2)

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

PBX #2

— Expects to receive three digits on trunk (D), and then uses these digits to route calls to lines (A).

— Routes calls starting with 0-7 to lines (A). — Routes calls starting with 0-7 to lines (B).

— Routes calls starting with 82 to trunk (G), and then the VoIP network uses these digits to route calls to trunk (E).

— Expects to receive three digits on trunk (E), and then uses these digits to route calls to lines (B).

— Routes calls starting with 81 to trunk (H), and then the VoIP network uses these digits to route calls to trunk (D).

Page 3-25

VoIP Networks without PSTN — Example 10

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — two hunt groups (24 channels per group/T1) (F)

Site preﬁx — single or multiple digits (J)

Voice phone group type — three digit local extensions (K)

Strip digit length — 2 (T)

Digital Interface type — T1 (W)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group site prefix digits (page 5-239)

voice phone group type (page 5-240) voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-26

VoIP Networks without PSTN — Example 11

Trunk Groups and Eleven Digit Extensions

This dialing scheme can be used with two voice daughtercards to translate seven digit trunk prefixes and four digit extensions. The seven digit site prefix, rather than the four digits extensions, are unique across the VoIP network. This enables each PBX site to handle the same or overlapping phone extensions.

The site prefix digits are used to send the VoIP calls to the correct PBX node. When a caller dials a specific site prefix, e.g., 1-603-598, it routes the call to the corresponding PBX and then dials a prefix to get a specific trunk. The user would dial 1-603-598-2xxx to call an extension off of PBX #2, for example. The VoIP daughtercard will strip the first seven digits and forward the last four digits to the PBX.

Any number used as a site prefix cannot be used for the first digit of any valid extension.

stripped

digits

(1-818-878) + (2000 to 2999)

(D)

Daughtercard

IP 192.168.11.2

(G)

Port 1720

Voice

Extensions

off of PBX #1

20002999

PBX #1

(A)

Extensions

off of PBX #2

20002999

PBX #2

(B)

stripped

digits

(1-603-598) + (2000 to 2999)

(H)

(E)

Daughtercard

IP 192.168.12.2

Port 1720

Example 11— Trunk Groups and Eleven Digit Extensions

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on trunk (D), and then uses these digits to route calls to lines (A).

— Routes calls starting with 2 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 1-603-598 to trunk (G), and then the VoIP network uses these digits to route calls to trunk (H).

— Expects to receive four digits on trunk (H), and then uses these digits to route calls to lines (B) or trunk (E).

— Routes calls starting with 1-818-878 to trunk (E), and then the VoIP network uses these digits to route calls to trunk (D).

WAN

Voice

Page 3-27

VoIP Networks without PSTN — Example 11

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — two hunt groups (24 channels per group/T1) (F)

Site preﬁx — single or multiple digits (J)

Voice phone group type — four digit local extensions (L)

Voice phone group type — eleven digit local extensions (M)

Strip digit length — 7 (V)

Digital Interface type — T1 (W)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group type (page 5-240) voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-28

VoIP Networks without PSTN — Example 12

H.323 Gatekeeper

This dialing scheme is used to connect voice switching daughtercards to an external H.323 NT100 RADVision (or other third-party) gatekeeper software application installed on a server or workstation. The phone groups for the individual daughtercard must be associated with the card in order for it to generate the H.323 alias telephone numbers for the gatekeepers. Configure the AVNDS for the voice daughtercards so that the H.323 destination is the gatekeeper instead of Trunks D or E. Most third-party gatekeepers should be compatible with the Alcatel’s H.323 gateway providing the gatekeepers are also H.323 compliant.

Extensions

off of PBX #1

10001999

Extensions

off of PBX #2

20002999

(A)

(B)

PBX #1

PBX #2

1000 to 1999

2000 to 2999

(D)

T1 T1

(G)

Gatekeeper (Zone 0)

H.323 Destination Alias

“1000”

“1001”

“1 . . .”

“1999”

“2000”

“2001”

“2 . . .”

“2999”

IP Address

192.168.11.2 {VSD #1)

192.168.12.2 {VSD #1)

192.168.12.2 {VSD #2)

IP . . . 13.2 Port 1719

(E)

(H)

Voice

Daughtercard

IP 192.168.11.2

Port 1720

Voice

Daughtercard

IP 192.168.12.2

Port 1720

WAN

Example 12 — H.323 Gatekeeper

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on trunk (D) and then uses these digits to route calls to lines (A).

— Routes calls starting with 1 to lines (A). — Routes calls starting with 2 to lines (B).

— Routes calls starting with 2 to trunk (G). The VoIP network then routes the call to the gatekeeper, and the gatekeeper uses these digits to route calls to trunk (E).

— Expects to receive four digits on trunk (E) and then uses these digits to route calls to lines (B).

— Routes calls starting with 1 to trunk (H). The VoIP network then routes the call to the gatekeeper, and the gatekeeper uses these digits to route calls to trunk (D).

Page 3-29

VoIP Networks without PSTN — Example 12

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

H.323 gateway to H.323 gatekeeper (RADVision) (B)

Local channel — two hunt groups (24 channels per group/T1) (F)

Site preﬁx — no site preﬁx (I)

Voice phone group type — four digit local extensions (L)

Strip digit length — no strip digits (R)

Digital Interface type — T1 (W)

voice network card h.323 gatekeeper control (page 5-217) voice network h.323 gatekeeper mode (page 5-218) voice network h.323 gatekeeper address (page 5-219) voice network h.323 gatekeeper associate (page 5-223)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

voice phone group site prefix (page 5-238) voice phone group type (page 5-240)

voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-30

VoIP Networks with PSTN — Example 13

North American PSTN and VoIP Calls

This dialing scheme is used for calls going through the North American PSTN and the VoIP Network. The following four diagrams are used to demonstrate how these calls are handled:

• North American PSTN Calls — Overview

• North American PSTN Calls — Outbound

• North American PSTN Calls — Inbound

• North American PSTN Calls — VoIP Network

Note that DID (Direct Inward Dial) is used only on inbound calls, and that all calls are connected using the North American Numbering Plan (NANP) that also includes Canada. See Dialing Scheme Example 15 for details on North American and International calls going through the PSTN or the VoIP network.

To use this dialing scheme, the voice phone group type must be set to NANP extensions. The site prefix digits and strip digits must be set to seven digits. NANP PSTN numbers depend on the configuration of the local exchange, of which there are three types: 1) those that support 7 digit dialing to get calls across the street (nxx-xxxx), and 2) those that support 10 digit dialing to get calls across the street (npa-nxx-xxxx) where NPA is the area code, and 3) those that support 11 digit dialing to get calls across the street (1-npa-nxx-xxxx) where npa is the area code.

When using voice phone group type NANP PSTN, the PSTN functionality is supported with the following exceptions:

• DID extensions (add numbers) and the NANP PSTN cannot be the same numbers.

• DID extensions (add numbers) and the site prefix digits cannot be the same numbers.

• Site prefix digits and NANP PSTN numbers cannot be the same numbers.

The phone group type NANP PSTN may be substituted by using site prefixes (forces callers to dial one or more digits, e.g., 9, to place an external call. For more information on using site prefixes, see also VoIP Networks without PSTN — Example 10 on page 3-25.

♦ Cautions ♦

Voice daughtercards using either voice phone group type of NANP PSTN or International PSTN dialing schemes cannot handle PBXs with extensions starting with 0, 1, 411 or 911. The following extensions are also not allowed: 0000 to 0999, 1000 to 1999, 4110 to 4119, and 9000 to 9999. (For more details, see Chapter 5, “VoIP Commands.”)

Companies using Alcatel’s VoIP feature are responsible for programming and testing all dialing schemes to reduce the likelihood or to eliminate the possibility of toll fraud from the PSTN.

Page 3-31

VoIP Networks with PSTN — Example 13

North American PSTN Calls — Overview

In this dialing scheme two voice switching daughtercards are used to translate area codes and telephone numbers. All eleven digits in the telephone numbers are unique across the VoIP network, and the voice switching daughtercards are responsible for all telephone number routing. 411 and 911 calls can be handled as well using this dialing scheme. Calls that are not routed across the VoIP network will be dropped and inserted, referred to as “Drop and Insert,” into the PSTN. Minimal to no PBX re-configuration is required; however, due to less than 99.995% reliability of Voice over IP networks, this dialing scheme is not recommended unless “passthrough” is used on some channels. (Passthrough not available this release).

To call a 2000 extension off of PBX#1, the caller dials an eleven digit NANP telephone number. In the overview diagram below, the voice daughtercard strips off the first seven digits, and then forwards the last four digits of the dialed number.

Extensions

off of PBX #1

2999

20002998

(A)

N. Amer.

0 . . . 411, 911

1-npa . . .

PBX #1

1-818-878

0 . . .

411, 911, 1-npa . . .

2000-2999

(R)

PSTN

(J)

(R)

Voice

Daughtercard

IP 192.168.11.2

Port 1720

stripped

digits

1-xxx-xxx

WAN

Intl.

011 . . . 001 . . .

ISDN PSTN

(T)

See Diagrams:

No. Amer. Calls

(Outbound) (Inbound DID) (VoIP Network)

1-xxx-xxx

stripped

digits

PBX #2

Extensions

off of PBX #2

2999

20002998

(B)

603-598

2000-2999

0 . . .

411, 911, 1-npa . . .

(T)

(L)

Voice

Daughtercard

IP 192.168.12.2

Port 1720

Example 13 — North American PSTN Calls (Overview)

LEGEND for Diagram Components

PBX #1 Conﬁguration PBX #2 Conﬁguration

— Expects to receive four digits on trunk (J) and then uses these digits to route calls to lines (A).

— Routes all calls starting with 2 to lines (A). — Routes all calls starting with 2 to lines (B).

— Expects to receive four digits on trunk (L) and then uses these digits to route calls to lines (B).

Page 3-32

VoIP Networks with PSTN — Example 13

LEGEND for Diagram Components

— Routes all 0 . . ., 411, 911 and 1-npa . . . calls to trunk (J), and then the VoIP network uses these digits to route calls to trunk (L), or trunk (R) if non-PSTN extensions.

— Routes all 0 . . ., 411, 911 and 1-npa . . . calls to trunk (L), and then the VoIP network uses these digits to route calls to trunk (J), or trunk (T) if non-PSTN extensions.

Remarks

Supported VoIP features and main CLI commands used with this dialing scheme are as follows, and are applicable to the outbound, inbound DID and North American VoIP calls.

Features Supported Primary CLI Commands Used

H.323 gateway to voice daughtercard (A) voice destination local channel (page 5-231)

Local channel — two hunt groups (24 channels per group/T1) (F)

voice numbering plan hunt method (page 5-255) voice numbering plan destination member (page 5-257) voice numbering plan phone group member (page 5-258)

Site preﬁx — single or multiple digits (J)

voice phone group site prefix (page 5-238) voice phone group site prefix digits (page 5-239)

Voice phone group type — NANP extensions (N)

voice phone group type (page 5-240) voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

Voice phone group type — PSTN NANP

(P)

voice phone group type (page 5-240) voice phone group format (page 5-243) voice phone group add numbers (page 5-247)

Strip digit length — 7 (V)

Digital Interface type — T1 (W)

voice phone group strip digit length (page 5-244) voice port interface type (page 5-34)

Page 3-33

VoIP Networks with PSTN — Example 13

North American PSTN Calls — Outbound

This diagram demonstrates how outbound North American PSTN calls are sent to the PSTN.

Extensions

off of PBX #1

2999

20002998

1-818-555-3001

PBX #1

(A)

1-818-878

N. Amer.

0 . . . 411, 911 1-npa . . .

1-603-598-2000

to 598-2999

0 . . . 411, 911

1-1-npa-nxx-xxxx

(R)

PSTN

See Diagrams:

No. Amer. Calls

(Overview) (Inbound DID) (VoIP Network)

1-800-555-9001

(J)

(R)

Example 13 — North American PSTN Calls (Outbound)

LEGEND for Diagram Components

Voice

Daughtercard

IP 192.168.11.2

Port 1720

WAN

PBX #1 Conﬁguration Voice Daughtercard #1 Conﬁguration

— Routes all calls to trunk (J) and sends four digits to voice daughtercard. See next diagram (No. Amer. PSTN Inbound) trunk (J) information.

— Expects to receive one to 24 digits on trunk (J), and then uses these digits to route calls to either trunk (R) or the VoIP network.

Page 3-34

VoIP Networks with PSTN — Example 13

North American PSTN Calls — Inbound DID (Direct Inward Dial)

This diagram demonstrates how inbound North American DID calls from the PSTN are handled.

PBX #1

Extensions

off of PBX #1

2999

20002998

(A)

1-818-878

(R)

PSTN

Example 13 — North American PSTN Calls (Inbound DID)

LEGEND for Diagram Components

PBX #1 Conﬁguration Voice Daughtercard #1 Conﬁguration

— Expects to receive four digits on trunk (J) and then uses these digits to route calls starting with 2 to lines (A). See previous diagram (No. Amer. PSTN Outbound) trunk (J) information.

Voice

2000-2999

(J)

Daughtercard

IP 192.168.11.2

Port 1720

(R)

See Diagrams:

No. Amer. Calls

(Overview) (Outbound) (VoIP Network)

WAN

— Expects to receive four digits on trunk (R), and then uses these digits to route calls to either trunk (J) or VoIP network.

Page 3-35

VoIP Networks with PSTN — Example 13

North American PSTN Calls — VoIP Network

This diagram demonstrates how North American calls are handled in a VoIP network.

When a 1-818-878-2000 extension is called from a 1-603-589-2000 extension, the PBX routes calls to Trunk J and then sends 1-603-598-2000 number to Voice Daughtercard #1.

The Voice Daughtercard #1 strips the 1st seven digits and forwards the last four digits across the WAN to Voice Daughtercard #2. These four digits are then forwarded to Trunk L. PBX #2 receives the four digits and then routes the call to the appropriate extension.

Extensions

off of PBX #1

2999

20002998

Extensions

off of PBX #2

2999

20002998

(A)

(D)

PBX #1

1-818-878

ISDN

PSTN

PBX #2

1-603-598

(R)

PSTN

(T)

0 . . . 411, 911

2000-2999

0 . . . 411, 911

(J)

(R)

See Diagrams:

No. Amer. Calls

(Overview) (Outbound) (Inbound DID)

(T)

(L)

Voice

Daughtercard

IP 192.168.11.2

Port 1720

stripped

digits

1-xxx-xxx

stripped

digits

Voice

Daughtercard

IP 192.168.12.2

Port 1720

WAN

Example 13 — North American PSTN Calls (VoIP Network)

LEGEND for Diagram Components

Voice Daughtercard #1 Conﬁguration Voice Daughtercard #2 Conﬁguration

— Expects to receive one to 24 digits on trunk (J) and then uses these digits to route calls to voice daughtercard #2.

— Routes all 1-603-598 calls to trunk (L). — Routes all 1-818-878 calls to trunk (J).

— Routes all 1-npa-nxx-xxxx PSTN calls to trunk (R). — Routes all 1-npa-nxx-xxxx PSTN calls to trunk (T).

— Routes all PSTN calls to trunk (R). — Routes all PSTN calls to trunk (T).

Page 3-36

— Expects to receive one to 24 digits on (L) and then uses these digits to route calls to voice daughtercard #1.

Alcatel-Lucent OMNIACCES-VOIP GATEWAY User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

FCC Compliance

Table of Contents

1 VoIP Overview

Introduction

VoIP Networks

Getting Started with VoIP

VoIP Telephone Calls

A VoIP Call Scenario

Elements of a Converged Network

VoIP H.323 Client

VoIP H.323 Gateway

VoIP H.323 Gatekeeper (Optional)

H.323 VoIP Gateway Voice and Convergence Features

Signaling Control and Voice Interoperability (Voice Features)

H.323 Call Control and Network Interoperability (Convergence Features)

H.323 Network Call Control

Alcatel VoIP Network Dialing Schemes (AVNDS)

Switch Backplane Interface

VoIP Standards for Development

Codec Support (G.711, G.723.1, G.729a)

VON (Voice on the Net) Developments

VoIP and VLANs

2 VoIP Daughtercards

Introduction

VoIP Daughtercard Types

Digital Signal Processors (DSPs), DIMMs and Available Channels

Voice Switching Daughtercard — Digital

VSD Front Panel

VSD Deadman Switch

VSD Cross-Over Toggle Switch

VSD Pinouts

VSD Jumpers

Voice Switching Daughtercard — Euro BRI ISDN

Digital Signal Processors (DSPs) and Available Channels

VSB Deadman Switch

VSB NT (LT)/TE Cross-Over Toggle Switch

VSB Pinouts

VSB Front Panel

VSB Jumpers

Voice Switching Daughtercard — Analog

VSA Front Panel

VSA Pinouts

VSAs and Digital Signal Processors (DSPs), DIMMs and Available Channels

VSAs and the Deadman Switch

VSAs and Cross-Over Toggle Switches

VSA Jumpers

VSX Switching Module

VoIP Daughtercard Port Numbering Schemes

Introduction

How to Select a Network Dialing Scheme (AVNDS)

Network Dialing Scheme VoIP Features

VoIP Networks without PSTN — Example 1

VoIP Networks without PSTN — Example 2

VoIP Networks without PSTN — Example 3

VoIP Networks without PSTN — Example 4

VoIP Networks without PSTN — Example 5

VoIP Networks without PSTN — Example 6

VoIP Networks without PSTN — Example 7

VoIP Networks without PSTN — Example 8

VoIP Networks without PSTN — Example 9

VoIP Networks without PSTN — Example 10

VoIP Networks without PSTN — Example 11

VoIP Networks without PSTN — Example 12

VoIP Networks with PSTN — Example 13