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Cisco Systems Linksys PAP2, RT31P2 6.5. Provisioning and Upgrade result codes , 6.6. Table of SIP Response Codes (Error Codes)

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X40 General SIP Protocol Error (e.g., unacceptable codec in SDP in 200 and

ACK messages, or times out while waiting for ACK)

X60 Dialed number invalid according to given dial plan

6.5. Provisioning and Upgrade result codes

The $PRVST and $UPGST macro variables expand to integer codes which report the state of a

resync or upgrade attempt. They are typically used within triggers and resync/upgrade conditions.

The values of these variables is as follows:

-1 = explicit request (resync/upgrade url or sip)

0 = just rebooted (resync only)

1 = triggered from configured trigger or rule

2 = error retry

6.6. Table of SIP Response Codes (Error Codes)

For convenience, below is a list of SIP error codes at the time of this printing which incorporates

response codes from the IANA (Internet Assigned Numbers Authority) SIP parameter registry

(http://www.iana.org/assignments/sip-parameters), and additional response codes defined in Internet-

drafts which are implemented by the PHONE ADAPTER.

Provisional 1xx

100 Trying

180 Ringing

181 Call Is Being Forwarded

182 Queued

183 Session Progress

Successful 2xx

200 OK

202 Accepted

Redirection 3xx

300 Multiple Choices

301 Moved Permanently

302 Moved Temporarily

305 Use Proxy

380 Alternative Service

Request Failure 4xx

400 Bad Request

401 Unauthorized

402 Payment Required

403 Forbidden

404 Not Found

405 Method Not Allowed

406 Not Acceptable

407 Proxy Authentication Required

408 Request Timeout

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

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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