Pstn | Cisco Systems ATA186I2ARF manual

Chapter 1 Cisco Analog Telephone Adaptor Overview

Session Initiation Protocol (SIP) Overview

Figure 1-2 Cisco ATA 186 as Endpoint in SIP Network

				Layer 3
Telephone/fax	V		Broadband	IP infrastructure
	V	Ethernet
	Cisco ATA 186		Broadband CPE
			(DSL, cable,
			fixed wireless)

SIP proxy

Voice

gateway

V	PSTN

72088

Figure 1-3 Cisco ATA 188 as Endpoint in SIP Network

				Layer 3
Telephone/fax	V		Broadband	IP infrastructure
	V	Ethernet
	Cisco ATA 188		Broadband CPE
			(DSL, cable,
			fixed wireless)

SIP proxy

Voice

gateway

V	PSTN

72444

Session Initiation Protocol (SIP) Overview

Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for real-time calls and conferencing over Internet Protocol (IP). SIP is an ASCII-based, application-layer control protocol (defined in RFC3261) that can be used to establish, maintain, and terminate multimedia sessions or calls between two or more endpoints.

Like other Voice over IP (VoIP) protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.

Note SIP for the Cisco ATA is compliant with RFC2543.

This section contains the following topics:

•SIP Capabilities, page 1-3

•Components of SIP, page 1-3

Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)

1-2	OL-3410-01

Image 20

Cisco Systems ATA186I2ARF manual Session Initiation Protocol SIP Overview, Pstn

Contents

Corporate Headquarters Text Part Number OL-3410-01 Copyright 2003, Cisco Systems, Inc All rights reserved N T E N T S Iii What You Need Dial Plan Disabling Access To The Web Interface Dhcp DNS1IP DNS2IP Vii Action Identifiers Viii Debugging Environmental Specifications C-2 Preface OverviewAudience This preface includes the following sections Organization ConventionsChapter Description Xii Xiii Related Documentation Xiv Obtaining Documentation World Wide WebDocumentation CD-ROM Ordering Documentation Obtaining Technical Assistance Cisco.comTechnical Assistance Center Xvi Xvii Cisco TAC Web SiteCisco TAC Escalation Center Xviii Cisco Analog Telephone Adaptor Overview Cisco ATA Analog Telephone Adaptor Session Initiation Protocol SIP Overview Pstn SIP Capabilities Components of SIP SIP Clients SIP Servers Hardware Overview Link LED Function Button Software Features Voice Codecs Supported Other Supported Protocols Additional Supported Signaling ProtocolsCisco ATA SIP Services Methods Supported Fax ServicesRegister Invite BYE Cancel Installation and Configuration Overview Supplementary ServicesNotify Options ACK Action Reference Installing the Cisco ATA What the Cisco ATA Package Includes Safety RecommendationsNetwork Requirements What You Need Installation Procedure5V power adaptor Power cord Procedure Cisco ATA 186 Rear Panel Connections Installing the Cisco ATA Installation Procedure Power-Down Procedure Configuring the Cisco ATA for SIP Default Boot Load Behavior Parameter and Bits Reference OpFlags VLANSetting FeatureBits Example Steps Needed to Configure the Cisco ATA Basic Configuration Steps in a Tftp Server Environment Basic Configuration Steps in a Non-TFTP Server Environment Configure the method with which the Cisco ATA will locateTftp server at boot up time Action Configurable Features and Related Parameters Configuring the Cisco ATA Using a Tftp ServerSetting Up the Tftp Server with Cisco ATA Software Creating Unique and Common Cisco ATA Configuration Files Save this file of Cisco ATA-specific parameters as Syntax Using atapname.exe Tool to Obtain MAC Address Command Example Syntax examples Using the EncryptKey Parameter and cfgfmt ToolCommand Output Atadefault.cfg Configuration File Using a Dhcp Server Using a Dhcp Server, Other Dhcp Options You Can Set DNS1IP DNS2IP Ntpip Voice Configuration MenuWithout Using a Dhcp Server Dhcp Using the Voice Configuration MenuVoice Menu Number Features Entering Alphanumeric Values Voice Menu Number Key Alphanumeric Characters Cisco ATA Web ConfigurationResetting the Cisco ATA to Factory Default Values Page UID0 Refreshing or Resetting the Cisco ATA Procedure to Reset the Cisco ATA Upgrading the SIP Signaling ImageProcedure to Refresh the Cisco ATA Required Parameters Basic and Additional SIP ServicesImportant Basic SIP Services Establishing Authentication Related Configuration Parameters Configuring Refresh Interval Additional SIP ServicesSetting the Codec Related Parameters Advanced Audio ConfigurationBillable Features Comfort Noise During Silence Period When Using G.711 Configurable Hook Flash TimingConfigurable Mixing of Call Waiting Tone and Audio Configurable On-hook delay Disabling Access To The Web Interface Dial PlanDistinctive Ringing DNS SRV Support NAT/PAT Translation Hardware Information DisplayNAT Gateway Progress Tones Network TimingOutBoundProxy Support Repeat Dialing on Busy Signal Receiver-tagged VIA header User Configurable Call Waiting Permanent Default Setting User Configurable Timeout On No Answer for Call ForwardingSIP Proxy Server Redundancy Stuttering Dial Tone on Unconditional Call Forward Setting Up and Placing a Call Without Using a SIP Proxy ConfigurationTo deactivate this feature, enter the following Open your Web browser Dial Format Complete Reference Table of all Cisco ATA SIP ServicesPlacing an IP Call OL-3410-01 Configurable Feature Related Parameter Configurable Feature Related Parameter OL-3410-01 Parameters and Defaults Configuration Text File Template UIPassword User Interface UI ParameterSections that follow describe these parameters Range Configuration Parameter Parameters for Configuration MethodToConfig UseTFTP CfgInterval Range Default Voice Configuration Menu Access CodeTftpURL Network Parameters EncryptKey IP address StaticIpStaticRoute 255.255.255.0 Account Information ParametersStaticNetMask This parameter is the password for the Phone 1 port UID0 This parameter is the password for the Phone 2 port GkOrProxy UseLoginID GatewayGateway2 Not applicable to SIP This parameter is the Login ID for line LoginID0LoginID1 Backup Server Parameters AltGkTimeOutAltGk Integer SIP Parameters GkTimeToLiveGkId UseSIP 205 SIPRegIntervalMAXRedirect SIPRegOn Natip OutBoundProxy SIPPortMediaPort Is the default port if no port is specified NatServerIP address or Fqdn format 207 NatTimer Operating ParametersBitmap 208 LBRCodec AudioMode300 0x00150015 RxCodec Bit Number Definition TxCodec NumTxFrames CallFeatures Clipclir First calling that number and consulting with the callee PaidFeatures315 CallerIdMethod Examples 316 FeatureTimer0x00019e60 Polarity 0x00000000317 304 ConnectMode RTP Bit Number Definition AutMethod TimeZoneThis parameter is not used for SIP 302 916 AltNTPIP142 SigTimer 9170xB8 0x01418564 Default 100 0x64=100 seconds Default-5 secondsRange 0 to 255 in 0.1 seconds Range 0 to 62 in seconds OpFlags 0x2 Optional Feature Parameters VLANSetting0x0000002b 324 Extended IP address NPrintfTraceFlags Default Recommended Values RingOnOffTime IPDialPlan DialPlan About Dial Plan Commands Additional DialPlan Information Following dial plan Following dial plans Dial Plan Blocking In Rule Rule to Support Hot/Warm Line Call-Progress Tone Parameters Rule to Support Dial PrefixList of Call-Progress Tone Parameters Tone Parameter Syntax How to Calculate Scaling Factors Use the following formula to calculate the scaling factor a Specific Call-Progress Tone Parameter Information Default values for the nine-integer arrayRecommended Values Cisco ATA plays the busy tone when the callee is busy On-time-0 Off-time-0 Total time to play tone-1000920 921 923 Off-time-2000 Total time to play tone-0922 CallCmd Off-time-2400 Total time to play tone-4800924 925 Maximum of 248 characters 930 Call Commands Call Command Structure Syntax CallCmd string has the following structure Context-Identifiers Identifier Context State of Cisco ATA Input Sequence Identifiers Action IdentifiersIdentifier Input Sequence Identifier Action Call Command Example Hook-flash Cancel-the-call-attempt Retrieve-the-waiting-call Call Command Behavior Call Command DefaultSweden Call Command Default Table Notations Call Command Behavior WFE cancels the call-Stop CWT and revert to Connected state CWT Call Command Behavior Call Command Behavior OL-3410-01 Configuring and Debugging Fax Services Using Fax Pass-through Mode Configuring the Cisco ATA for Fax Pass-through mode This setting translates to the following bitmapAudioMode Translation Configuring Cisco IOS Gateways to Enable Fax Pass-through Recommended SettingThis setting translates to the bitmap ConnectMode Enable Fax Pass-through Mode Run the following command Disable Fax Relay Feature Perform the command Using FAX Mode Configuring the Cisco ATA for Fax Mode Debugging the Cisco ATA 186/188 Fax Services Configuring the Cisco ATA for Fax Mode on a Per-Call BasisConfiguring the Cisco IOS Gateway for Fax Mode Common Problems When Using IOS Gateways Problem Action Cisco ATA, and 0x0012XXXX for the Phone 2 portFor fax pass-through mode, AudioMode should be set to Port Analyzing prserv Output for Fax Sessions Using prserv for Diagnosing Fax ProblemsPrserv Overview Log event Description Terminating-Gateway Example Originating-Gateway Example Possible Reasons for Failure To use rtpcatch, follow these steps Using rtpcatch for Diagnosing Fax ProblemsRtpcatch Overview Example of rtpcatch Output Files Explanation CED tone Detected Fax relay mode Cisco fax relay mode Analyzing rtpcatch Output for Fax SessionsAnalysis Both sides use G.711 for the entire fax session Using rtpcatch to Analyze Common Causes of Failure Example 7-4 T38 Fax Relay Mode Possible Cause for Failure Cisco fax relay option is not disabled on the gatewayPossible Causes for Failure Example 7-9 Fax Pass-through Mode Failure Rtpcatch Limitations OL-3410-01 Upgrading the Cisco ATA Signaling Image Upgrading the Signaling Image from a Tftp ServerSyntax of upgradecode Parameter Definitions Process Upgrading the Signaling Image ManuallyUpgradecode parameter value could be Upgrade Requirements Preliminary StepsRunning the Executable File Syntax Upgrade Procedure To perform the upgrade, follow these steps Confirming a Successful Signaling Image Upgrade Using a Web Browser OL-3410-01 Troubleshooting General Troubleshooting Tips Symptoms and Actions Installation and Upgrade Issues Debugging 2* 1 1* 2* 8* 1 1* 7* 8* 1 1* 9* 0* 1 1* 9* 0* 0 Frequently Asked Questions FACTRESET#Ring Load per RJ-11 FXS Port Maximum Distance REN Contacting TAC Using SIP Supplementary Services Changing Call CommandsCancelling a Supplementary Service Common Supplementary Services Voice Mail Indication Caller IDCall-Waiting Caller ID Fully Unattended Transfer Unattended TransferSemi-unattended Transfer Making a Conference Call in Sweden Attended TransferMaking a Conference Call in the United States Call Waiting in the United States Call Waiting in SwedenAbout Call Forwarding Call Forwarding in the United States Call Forwarding in Sweden Call Return in the United StatesCall Return in Sweden Calling Line Identification Presentation Calling Line Identification Restriction in the United States Calling Line Identification Restriction in Sweden OL-3410-01 Voice Menu Option Code Description Voice Menu CodesTable B-1lists codes to return basic Cisco ATA information PWD DNS 1 IPDNS 2 IP Tftp URL UIDUser ID telephone number for the Phone 1 port User ID telephone number for the Phone 2 port Option Code Description Specification Cisco ATA SpecificationsPhysical Specifications Electrical Specifications Environmental SpecificationsImmunity Specifications Description Specification Physical Interfaces Ringing CharacteristicsSoftware Specifications Appendix C Cisco ATA Specifications Software Specifications SIP Compliance Reference Information Sccp OL-3410-01 SIP Call Flows Supported SIP Request Methods Step Action Description Cisco ATA-to-SIP Server-Registration without AuthenticationCall Flow Scenarios for Successful Calls Cisco ATA-to-SIP Server-Registration with Authentication Table D-2 Log Listings Table D-3 Action Descriptions Table D-4 Log Listings Invite Table D-5 Action Descriptions Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Trying Proxy Authentication Required ACK Table D-7 Action Descriptions Table D-8 Log Listings Table D-8 Log Listings Table D-8 Log Listings Table D-8 Log Listings Table D-8 Log Listings GL-1 GL-2 GL-3 Lightweight DirectoryAccess Protocol Location services are offered by location serversLocal exchange carrier Media Gateway Control Protocol GL-5 Signaling connection control partMessages can be part of Sgcp and Mgcp messages GL-6 Business-class services for Internet telephony TrafficAllow you to define your own customized markup language GL-7 GL-8 D E IN-1 IN-2 United States Call-progress tone parametersCfgInterval parameter IN-3 Fax mode configuration 7-6,7-7 IN-4 Forward-on-no-answer setting Forward-unconditionally settingFXS ports IN-5 IN-6 NatipNatip parameter RJ-45 LED IN-7 SIP registration SIP registration enabled IN-8 Unconditional call forwarding Unique configuration file IN-9 IN-10