Linksys SPA-1001 Manuel utilisateur

Corporate Headquarters

Linksys

121 Theory Drive

Irvine, CA 92617

USA

http://www.linksys.com

Tel: 949 823-1200

800 546-5797

Fax: 949 823-1100

Linksys ATA Administrator Users Guide

Document Version 3.2

Linksys ATA Administrator Guide



Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and product names are trademarks or registered trademarks of their respective holders.

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Linksys ATA Administrator Guide

Document Version 3.1

CONTENTS

Preface

Document Audience i-xi

Linksys Analog Telephone Adapters i-xi

How This Document is Organized i-xii

Document Conventions i-xii

Related Documentation i-xiii

Technical Support i-xiii

i-xiii

CHAPTER

1Introducing Linksys Analog Telephone Adapters 1-1

Overview 1-1

Ensuring Voice Quality 1-3

Audio Compression Algorithm 1-4

Silence Suppression 1-4

Packet Loss 1-4

Network Jitter 1-4

Echo 1-4

Hardware Noise 1-5

End-to-End Delay 1-5

Feature Descriptions 1-5

SIP Proxy Redundancy 1-5

Supported Codecs 1-6

Streaming Audio Server and Music on Hold 1-6

Silence Suppression and Comfort Noise Generation 1-7

Modem and Fax Pass-Through 1-7

Adaptive Jitter Buffer 1-7

Other Features 1-8

Technology Background 1-9

Session Initiation Protocol 1-10

Network Address Translation 1-10

NAT Overview 1-10

NAT Types 1-11

Contents

Linksys ATA Administrator Guide

Document Version 3.1

Simple Traversal of UDP Through NAT 1-12

SIP-NAT Interoperation 1-12

Where to Go From Here 1-13

CHAPTER

2Getting Started 2-1

Linksys Analog Telephone Adapters (ATAs) 2-1

Caring for Your Hardware 2-2

AG310 2-2

PAP2T 2-3

RTP300 2-4

SPA1001 2-5

SPA2102 2-6

SPA3102 2-7

SPA8000 2-9

WRP400 2-10

WRTP54G 2-11

WRT54GP2 2-13

Establishing Connectivity 2-14

Bandwidth Requirements 2-14

Making the Physical Connections 2-15

Connecting the SPA8000 2-16

SPA8000 Architecture 2-16

Connectivity Requirements 2-17

Using the Interactive Voice Response Interface 2-18

Using the IVR Menu 2-18

IVR Options 2-19

Entering a Password through the IVR 2-21

Using the Administration Web Server 2-22

Connecting to the Administration Web Server 2-22

Administrator Account Privileges 2-23

CHAPTER

3Configuring Linksys ATAs 3-1

Initial Configuration 3-1

Web Interface URLs 3-3

Upgrade URL 3-3

Resync URL 3-3

Reboot URL 3-4

Provisioning 3-4

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Linksys ATA Administrator Guide

Document Version 3.1

Provisioning Capabilities 3-4

Configuration Profile 3-4

Configuring a Dial Plan 3-5

Dial Plan Digit Sequences 3-5

Dial Plan Rules 3-6

Digit Sequence Syntax 3-6

Element Repetition 3-7

Sub-sequence Substitution 3-7

Inter-sequence Tones 3-7

Number Barring 3-7

Interdigit Timer Master Override 3-7

Local Timer Overrides 3-7

Pause 3-7

Dial Plan Examples 3-8

Dial Plan Timers 3-9

Interdigit Long Timer 3-9

Interdigit Short Timer 3-9

Dial Plans 3-9

Secure Call Implementation 3-10

Enabling Secure Calls 3-10

Secure Call Details 3-10

Using a Mini-Certificate 3-11

Generating a Mini-Certificate 3-11

Configuring a Streaming Audio Server 3-12

Music On Hold 3-12

Using a Streaming Audio Server 3-13

Using the IVR with an SAS Line 3-13

Example SAS with MOH 3-14

SAS Line Registered with the Proxy Server 3-14

SAS Line Not Registered with the Proxy Server 3-14

Configuring the Streaming Audio Server 3-15

Using a FAX Machine with the SPA2102 or SPA8000 3-15

Managing Caller ID Service 3-17

Troubleshooting and Configuration FAQ 3-18

CHAPTER

4Configuring the PSTN Gateway (FXO) 4-1

Overview 4-1

How VoIP-To-PSTN Calls Work 4-2

One-Stage Dialing 4-2

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Linksys ATA Administrator Guide

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Two-Stage Dialing 4-3

How PSTN-To-VoIP Calls Work 4-4

Terminating Gateway Calls 4-4

VoIP Outbound Call Routing 4-5

Configuring VoIP Failover to PSTN 4-7

Sharing One VoIP Account Between the FXS and PSTN Lines 4-7

Other Options 4-7

PSTN Call to Ring Line 1 4-8

Symmetric RTP 4-8

Call Progress Tones 4-8

Call Scenarios 4-8

PSTN to VoIP Call with and Without Ring-Thru 4-9

VoIP to PSTN Call with and Without Authentication 4-9

Using PIN Authentication 4-9

Using HTTP Digest Authentication 4-10

Without Authentication 4-10

Call Forwarding to PSTN Gateway 4-10

Forward-On-No-Answer to the PSTN Gateway 4-11

Forward-All to the PSTN gateway 4-11

Forward to a Particular PSTN Number 4-11

Forward-On-Busy to PSTN Gateway or Number 4-11

Forward-Selective to PSTN Gateway or Number 4-11

User Dialing 9 to Access PSTN-Gateway for Local Calls 4-11

Using the PSTN-Gateway for 311 and 911 Calls 4-12

Auto-Fallback to the PSTN-Gateway 4-12

CHAPTER

5Linksys ATA Field Reference 5-1

Info Tab 5-2

System Information (PAP2T) 5-2

System Status (VoIP) 5-2

Line 1/2 Status 5-3

PSTN Line Status 5-5

System Tab 5-8

System Configuration 5-8

Internet Connection Type (PAP2T) 5-8

Optional Network Configuration (PAP2T) 5-9

Miscellaneous Settings (Not in PAP2T) 5-10

SIP Tab 5-11

SIP Parameters 5-11

Contents

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SIP Timer Values (sec) 5-12

Response Status Code Handling 5-14

RTP Parameters 5-14

SDP Payload Types 5-15

NAT Support Parameters 5-16

Regional Tab 5-19

Call Progress Tones 5-19

Distinctive Ring Patterns 5-21

Distinctive Call Waiting Tone Patterns 5-21

Distinctive Ring/CWT Pattern Names 5-22

Ring and Call Waiting Tone Spec 5-23

Control Timer Values (sec) 5-23

Vertical Service Activation Codes 5-24

Vertical Service Announcement Codes 5-28

Outbound Call Codec Selection Codes 5-28

Miscellaneous 5-30

Line Tab 5-33

Line Enable 5-33

Streaming Audio Server (SAS) 5-33

NAT Settings 5-34

Network Settings 5-35

SIP Settings 5-35

Call Feature Settings 5-38

Proxy and Registration 5-38

Subscriber Information 5-40

Supplementary Service Subscription 5-40

Audio Configuration 5-42

Gateway Accounts (SPA3102/AG310) 5-45

VoIP Fallback to PSTN (SPA3102/AG310) 5-46

Dial Plan 5-46

FXS Port Polarity Configuration 5-47

PSTN Line Tab 5-49

Line Enable 5-49

NAT Settings 5-49

Network Settings 5-50

SIP Settings 5-50

Proxy and Registration (SPA3102/AG310) 5-53

Subscriber Information (SPA3102/AG310) 5-54

Audio Configuration (SPA3102/AG310) 5-55

Contents

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Dial Plans 5-57

VoIP-To-PSTN Gateway Setup 5-57

VoIP Users and Passwords (HTTP Authentication) 5-59

Ring Settings 5-60

FXO (PSTN) Timer Values (sec) 5-60

PSTN Disconnect Detection 5-62

International Control (Settings) 5-63

User 1/2 Tab 5-65

Call Forward Settings 5-65

Selective Call Forward Settings 5-66

Speed Dial Settings 5-67

Supplementary Service Settings 5-67

Distinctive Ring Settings 5-68

Ring Settings 5-68

PSTN User Tab (SPA3102/AG310) 5-70

PSTN-To-VoIP Selective Call Forward Settings 5-70

PSTN-To-VoIP Speed Dial Settings 5-70

PSTN Ring Thru Line 1 Distinctive Ring Settings 5-70

PSTN Ring Thru Line 1 Ring Settings 5-71

PSTN/VoIP Caller Commands via DTMF 5-71

APPENDIX

AAcronyms

APPENDIX

BGlossary

APPENDIX

CUser Guidelines

Basic Services C-1

Originating a Phone Call C-1

Receiving a Phone Call C-2

Enhanced Services C-2

Caller ID C-2

Calling Line Identification Presentation (CLIP) C-2

Calling Line Identification Restriction (CLIR)—Caller ID Blocking C-3

Call Waiting C-3

Disable or Cancel Call Waiting C-4

Call-Waiting with Caller ID C-5

Voice Mail C-5

Attendant Call Transfer C-6

Unattended or “Blind” Call Transfer C-6

Contents

Linksys ATA Administrator Guide

Document Version 3.1

Call Hold C-7

Three-Way Calling C-7

Three-Way Ad-Hoc Conference Calling C-8

Call Return C-8

Automatic Call Back C-9

Call FWD—Unconditional C-9

Call FWD – Busy C-10

Call FWD—No Answer C-11

Anonymous Call Blocking C-11

Distinctive/Priority Ringing and Call Waiting Tone C-12

Speed Calling—Up to Eight Numbers or IP Addresses C-12

NDEX

Contents

Linksys ATA Administrator Guide

Document Version 3.1

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