Linksys SPA-1001 User manual

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Linksys ATA Administrator Users Guide
Document Version 3.2

Linksys ATA Administrator Guide 
Copyright ©2007 Cisco Systems, Inc. All rights reserved.Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems,
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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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

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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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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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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

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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

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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

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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
I NDEX

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