Cisco Systems Linksys PAP2, RT31P2 - page 71

Max Redirection Number of times to allow an INVITE to be

redirected by a 3xx response to avoid an

infinite loop.

Note: This parameter currently has no effect: there is

no limit on number of redirection.

Uns8 5

Max Auth Maximum number of times a request may be

challenged (0-255) Uns8 2

SIP User Agent

Name User-Agent Header to be used by the unit in

outbound requests. If empty, the header is not

included.

Str63 Linksys/

$version

SIP Server Name Server Header to used by the unit in

responses to inbound responses. If empty,

the header is not included.

Str63 Linksys/

$version

SIP Accept

Language Accept-Language Header to be used by the

unit.

If empty, the header is not included.

Str31

Remove Last Reg Remove last registration before registering a

new one if value is different one. Bool no

Use Compact

Header If set to yes, the PHONE ADAPTER will use

compact SIP headers in outbound SIP

messages. If set to no the PHONE ADAPTER

will use normal SIP headers.

Bool no

SIP Timer Values (sec)

SIP T1 RFC 3261 T1 value (RTT Estimate). Range: 0

– 64 sec Time3 .5

SIP T2 RFC 3261 T2 value (Maximum retransmit

interval for non-INVITE requests and INVITE

responses). Range: 0 – 64 sec

Time3 4

SIP T4 RFC 3261 T4 value (Maximum duration a

message will remain in the network). Range:

0 – 64 sec

Time3 5

SIP Timer B INVITE time out value. Range: 0 – 64 sec Time3 32

SIP Timer F Non-INVITE time out value. Range: 0 – 64

sec Time3 32

SIP Timer H INVITE final response time out value. Range:

0 – 64 sec Time3 32

SIP Timer D ACK hang around time. Range: 0 – 64 sec Time3 32

SIP Timer J Non-INVITE response hang around time.

Range: 0 – 64 sec Time3 32

INVITE Expires INVITE request Expires header value in sec.

0 = do not include Expires header in INVITE.

Range: 0 – (231 – 1)

Time0 180

ReINVITE Expires ReINVITE request Expires header value in

sec. 0 = do not include Expires header in the

request. Range: 0 – (231 – 1)

Time0 30

Reg Min Expires Minimum registration expiration time allowed

from the proxy in the Expires header or as a

Contact header parameter. If proxy returns

something less this value, then the minimum

value is used.

Time0 1

Reg Max Expires Maximum registration expiration time allowed

from the proxy in the Min-Expires header. If Time0 7200

© 2004 Linksys Proprietary (See Copyright Notice on Page 2)

71

Contents

Main Linksys PAP2 and RT31P2 Disclaimer Please Read: Use of Proprietary Information and Copyright Notice: Page Page Page 1. Introduction 1.1. The Session Initiation Protocol Page 1.1.1. Components of a SIP Network Service Provider Domain Subscriber Domain IP Network (Internet) 1.1.2. Provisioning Overview 1.1.3. Security Overview 1.1.4. SIP Services Page Page Page 1.2. Network Address Translation (NAT) Traversal 1.2.1. What is a NAT or NAPT (Network Address Port Translator)? 1.2.2. VoIP-NAT Interworking 1.3. Voice Quality Overview 2. Hardware Overview Page 2.1. Phone Adapter LED Status 2.2. Broadband Router (RT31P2) LED Status 3. Software Configuration Mechanisms 3.1. Configuration Profile Formats Page 3.1.1. Using the Supplemental Profile Compiler 3.1.2. Encrypting and Compressing XML configuration files 3.2. Secure Initial Configuration 3.3. Web Interface 3.3.1. Web Interface Conventions 3.3.2. Administration Privileges 3.3.3. Basic and Advanced Views 3.4.1. Upgrade URL 3.4.2. Resync URL 3.4.3. Reboot URL 3.4.4. Factory Reset URL 3.5. Configuration via the IVR (PAP2 only) Page Page 4. Configuration Parameters 4.1. Data Types Page 2004 Linksys Proprietary (See Copyright Notice on Page 2) 4.1.1. Conventions 4.2. Provisioning Related Parameters Page Page Page Page Page Page Page 4.2.1. Firmware Upgrade Page Page 4.2.2. Provisioning Server Redundancy 4.2.3. Configuring the Web Server and IVR 4.3. Basic Networking Configuration 4.4. Basic Account Configuration 4.5. Configuration for NAT Traversal Page 4.6. Media and SDP (Session Description Protocol) Configuration 4.6.1. DTMF and Hookflash 4.6.2. Codec and Audio Settings 4.6.3. Dynamic Payload Types and SDP Codec Names 4.6.4. Secure Media Implementation: Page 4.6.5. Outbound Call Codec Selection Codes: 4.7. Supplementary Services 4.7.1. Supplementary Services activated internally Page 4.7.2. Call Forwarding Implemented internally 4.7.3. Supplementary Services implemented in the service provider network 4.8. Dial Plan Configuration Page Page Page Page 4.8.1. Speed Dialing Settings 4.9. Progress Tone and Ring Configuration 4.9.1. Distinctive Ring and Other Ring Settings Page 4.9.2. Progress Tones 4.10. Less Frequently Used Paramters Page Page 4.10.2. Additional User Account Information Page 4.10.3. Per-Line Polarity Settings 4.10.4. Additional Timer Values (sec) 4.10.5. Miscellaneous Parameters Page Figure: PHONE ADAPTER Caller ID Delivery Architecture 5. Expected Feature Behavior 5.1. Originating a Phone Call 5.2. Receiving a Phone Call 5.3. Caller ID 5.4. Calling Line Identification Presentation (CLIP) 5.5. Calling Line Identification Restriction (CLIR) Caller ID Blocking 5.6. Call Waiting 5.7. Disable or Cancel Call Waiting 5.8. Call-Waiting with Caller ID 5.9. Voice Mail 5.10. Attendant Call Transfer 5.11. Unattended or Blind Call Transfer 5.12. Call Hold 5.13. Three-Way Calling 5.14. Three-Way Ad-Hoc Conference Calling 5.15. Call Return 5.16. Automatic Call Back 5.17. Call FWD Unconditional 5.18. Call FWD Busy 5.19. Call FWD - No Answer 5.20. Anonymous Call Blocking 5.21. Distinctive / Priority Ringing and Call Waiting Tone 5.22. Speed Calling Up to Eight (8) Numbers or IP Addresses 6. Troubleshooting 6.1. Call Statistics Reporting 6.2. Enabling Logging and Debugging 6.3. Error and Log Reporting 6.4. Internal Error Codes 6.5. Provisioning and Upgrade result codes 6.6. Table of SIP Response Codes (Error Codes) 7. Summary of Implemented Features and Specifications 7.1. Data Networking Features 7.1.1. MAC Address (IEEE 802.3) 7.2. Voice Features 7.2.3. Secure Calls 7.2.4. Voice Algorithms: 7.2.5. Codec Selection 7.2.6. Dynamic Payload 7.2.7. Adjustable Audio Frames Per Packet 7.2.10. Call Progress Tone Generation 7.2.11. Call Progress Tone Pass Through 7.2.12. Jitter Buffer Dynamic (Adaptive) 7.2.13. Full Duplex Audio 7.2.14. Echo Cancellation Up to 8 ms Echo Tail 7.2.20. Message Waiting Indicator Tones MWI 7.2.21. Polarity Control 7.2.22. Calling Party Control CPC 7.2.23. International Caller ID Delivery 7.2.24. Streaming Audio Server SAS 7.2.25. Music On Hold MOH Page 7.3. Security Features dministration and Maintenance Features ers 7.4. A 8. List of all configuration paramet Page Page Page Page Page Page Page Page Page Page 9. Acronyms Page 10. Glossary