Cisco Systems ATA 188 manual GL-3

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FXO

FXS

Glossary

Foreign Exchange Office. An FXO interface connects to the public switched telephone network (PSTN) central office and is the interface offered on a standard telephone. Cisco FXO interface is an RJ-11 connector that allows an analog connection at the PSTN central office or to a station interface on a PBX.

Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.

G

G.711

Describes the 64-kbps PCM voice coding technique. In G.711, encoded voice is already in the correct

 

format for digital voice delivery in the PSTN or through PBXs. Described in the ITU-T standard in its

 

G-series recommendations.

G.723.1

Describes a compression technique that can be used for compressing speech or audio signal

 

components at a very low bit rate as part of the H.324 family of standards. This Codec has two bit

 

rates associated with it: 5.3 and 6.3 kbps. The higher bit rate is based on ML-MLQ technology and

 

provides a somewhat higher quality of sound. The lower bit rate is based on CELP and provides

 

system designers with additional flexibility. Described in the ITU-T standard in its G-series

 

recommendations.

G.729A

Describes CELP compression where voice is coded into 8-kbps streams. There are two variations of

 

this standard (G.729 and G.729 Annex A) that differ mainly in computational complexity; both

 

provide speech quality similar to 32-kbps ADPCM. Described in the ITU-T standard in its G-series

 

recommendations.

gateway

A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols

 

by converting protocols. A gateway is the point where a circuit-switched call is encoded and

 

repackaged into IP packets.

H

H.245

An ITU standard that governs H.245 endpoint control.

H.323

H.323 allows dissimilar communication devices to communicate with each other by using a standard

 

communication protocol. H.323 defines a common set of CODECs, call setup and negotiating

 

procedures, and basic data transport methods.

I

ICMP

Internet Control Message Protocol

Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (H.323)

 

OL-4008-01

GL-3

 

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Contents Customer Order Number OL-4008-01 Corporate HeadquartersCopyright 2003, Cisco Systems, Inc All rights reserved Iii N T E N T SSafety Recommendations ToConfig Dhcp Vii DNS1IP DNS2IPViii Using FAX ModeContacting TAC OL-4008-01 This preface includes the following sections OverviewAudience Xii OrganizationConventions Chapter DescriptionXiii Xiv Related DocumentationOrdering Documentation Obtaining DocumentationWorld Wide Web Documentation CD-ROMWe appreciate your comments Obtaining Technical AssistanceCisco.com Technical Assistance CenterXvii Cisco TAC Web SiteCisco TAC Escalation Center Xviii Cisco Analog Telephone Adaptor Overview Cisco ATA 186 as Endpoint in an H.323 Network Gateways TerminalsProxy Server GatekeepersMCUs Cisco ATA 186-Rear View Hardware OverviewFunction Button Other Supported Protocols Software FeaturesAdditional Supported Signaling Protocols Voice Codecs SupportedCisco ATA H.323 Services Action Reference Installation and Configuration OverviewFax Services Supplementary ServicesImage Upgrading the Cisco ATA SignalingImage by using the Tftp server-upgrade method or Manual-upgrade methodSafety Recommendations Installing the Cisco ATAWhat You Need Installation ProcedureWhat the Cisco ATA Package Includes Cisco ATA 186 Rear Panel Connections ProcedureInstalling the Cisco ATA Installation Procedure Power-Down Procedure OL-4008-01 Configuring the Cisco ATA for H.323 Default Boot Load Behavior OpFlags, VLANSetting,Parameter and Bits Reference OpFlagsExample VLANSettingFeature BitsUpgrading the Signaling Image from a Tftp Server Steps Needed to Configure the Cisco ATABasic Configuration Steps in a Tftp Server Environment Atadefault.cfg Configuration File,Refreshing or Resetting the Cisco ATA, Basic Configuration Steps in a Non-TFTP Server EnvironmentConfiguring the Cisco ATA to Obtain its Configuration File from the Tftp Server,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 Syntax Save this file of Cisco ATA-specific parameters asCommand Output Using atapname.exe Tool to Obtain MAC AddressCommand Example Syntax examples Using the EncryptKey Parameter and cfgfmt ToolWithout Using a Dhcp Server, Atadefault.cfg Configuration FileUsing a Dhcp Server, Using a Dhcp Server Other Dhcp Options You Can Set Without Using a Dhcp ServerDNS1IP DNS2IP Ntpip Voice Configuration MenuVoice Menu Number Features Using the Voice Configuration MenuKey Alphanumeric Characters Entering Alphanumeric ValuesWhere ipaddress is the IP address of the Cisco ATA Cisco ATA Web ConfigurationResetting the Cisco ATA to Factory Default Values UID0 OpFlags, page 5-33-Bit Web Interface Access-Control ConfigurationRefreshing or Resetting the Cisco ATA Related ParameterHttp Refresh and Reset Access-Control Configuration Upgrading the H.323 Signaling ImageHttp Procedure to Refresh the Cisco ATA Http Procedure to Reset the Cisco ATAOL-4008-01 Required Parameters Important Basic H.323 ServicesSetting the Signaling Image to H.323 Mode UID0, UID1, Setting Up User IDs for the Cisco ATAUsing the Cisco ATA with an H.323 Gatekeeper Related Configuration ParametersAltGk, AltGkTimeOut, ConnectMode, Setting Up Gatekeeper Time-To-Live ValueSetting Up an Alternate H.323 Gatekeeper LoginID0, LoginID1, UID0, UID1,Establishing Authentication with Cisco H.323 Gatekeeper Gateway, GkOrProxy, Using the Cisco ATA Without an H.323 GatekeeperUsing Multiple Cisco ATAs Without an H.323 Gatekeeper LBRCodec, AudioMode, Additional H.323 ServicesSetting the Audio Codecs IPDialPlan,Configuring Billable Features Configuring Audio Packet SettingsConfiguring the Mixing of Call Waiting Tone and Audio Configuring the Call Waiting Permanent Default SettingConfiguring the Cisco ATA Refresh Interval Configuring Hook Flash TimingDebugging Diagnostics Configuring On-hook delayConfiguring Reverse Audio Cut-Through Behavior Configuring Supplementary Service Behavior and ParametersSelecting Dtmf and Hookflash Transmission Methods Polarity SettingsNetwork Timing Progress TonesDialPlan, Setting Dial PlansSelecting H.323 Connection and H.245 Transmission Methods ConnectMode, page 5-28-Bits 0Configurable Features Related Parameters OL-4008-01 Parameters and Defaults Configuration Text File Template Sections that follow describe these parameters UIPasswordThis section contains only one parameter-UIPassword User Interface UI ParameterToConfig Configuration-Complete ParameterParameters for Configuration Method and Encryption TftpURL SettingsRange Default Voice Configuration Menu Access Code UseTFTPUseTFTP, TftpURL, CfgIntervalEncryptKey UseTFTP, CfgInterval,DHCP, StaticIp, StaticRoute, StaticNetMask, Network Parameters320 DHCP, StaticRoute, StaticNetMask, Voice Configuration Menu Access Code Related ParametersStaticIp StaticRoute255.255.255.0 Account Information ParametersStaticNetMask DHCP, StaticIp, StaticNetMask,This parameter is the password for the Phone 1 port UID1, PWD0, PWD1, UseLoginID, LoginID0, LoginID1,UID0, PWD0, PWD1, UseLoginID, LoginID0, LoginID1, UID0, UID1, PWD1, UseLoginID, LoginID0, LoginID1, AutMethod,Gateway This parameter is the password for the Phone 2 portLoginID1, PWD0, PWD1, UseLoginID, AutMethod, UseLoginIDLoginID0 Bitmap LoginID1LoginID0, PWD0, PWD1, UseLoginID, AutMethod, AutMethodGatekeeper Parameters LoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP,GkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId, AltGk, AltGkTimeOut, GkTimeToLive, GkId,30 to 4294967295 seconds AltGkTimeOutAltGk IntegerNot specified Default RangeGkTimeToLive GkIdUseSIP Mode ParameterUse H.323 mode -Use SIP mode Operating ParametersLBRCodec DNS2IP, UDPTOS, SigTimer, OpFlags, VLANSetting,UDPTOS, VLANSetting, MediaPortLBRCodec, ConnectMode, RxCodec, AudioModeRxCodec, TxCodec, AudioMode, page 5-20-Bits 1 TxCodec, RxCodec, NumTxFrames,Bit Number Definition RxCodecLBRCodec, NumTxFrames, RxCodec, AudioMode, TxCodecLBRCodec, NumTxFrames, TxCodec, AudioMode, LBRCodec, RxCodec, TxCodec, NumTxFramesExamples Bit Number CallFeatures315 PaidFeaturesCallFeatures, CallCmd, CallerIdMethod, SigTimer, CallerIdMethod 316 Polarity0x00019e60 ConnectMode Use G.711A-law for fax pass-through codec TimeZoneUse G.711µ-law for fax pass-through codec AltNTPIP, TimeZone, AltNTPIPNTPIP, AltNTPIP, 141917 NTPIP, TimeZone,916 SigTimer OpFlags TftpURL, DHCP, VLANSetting,VLANSetting 324 Optional Feature ParametersNPrintf 0x0000002bIPDialPlan Default Recommended ValuesRingOnOffTime Additional DialPlan Information DialPlanFollowing dial plan About Dial Plan CommandsFollowing dial plans Dial Plan Blocking In RuleRule to Support Dial Prefix Rule to Support Hotline/WarmlineEach tone is specified by nine integers, as follows Call-Progress Tone ParametersList of Call-Progress Tone Parameters Tone Parameter SyntaxUse the following formula to calculate the scaling factor a How to Calculate Scaling FactorsRecommended Values 920 Default values for the nine-integer arraySpecific Call-Progress Tone Parameter Information 922 Cisco ATA plays the busy tone when the callee is busy921 924 923Maximum of 248 characters CallCmd925 930 CallFeatures, PaidFeatures, CallerIdMethod, SigTimer,OL-4008-01 Call Command Structure Call CommandsCallCmd string has the following structure SyntaxIdentifier Context State of Cisco ATA Context-IdentifiersIdentifier Action Input Sequence IdentifiersAction Identifiers Identifier Input SequenceCall Command Example Hook-flash Cancel-the-call-attempt Retrieve-the-waiting-call Table Notations Call Command BehaviorCall Command Default Sweden Call Command DefaultCall Command Behavior CWT WFE cancels the call-Stop CWT and revert to Connected stateCall Command Behavior Call Command Behavior OL-4008-01 Using Fax Pass-through Mode Configuring and Debugging Fax ServicesAudioMode, ConnectMode, Configuring the Cisco ATA for Fax Pass-through modeThis setting translates to the following bitmap AudioModeEnable Fax Pass-through Mode, Disable Fax Relay Feature, Configuring Cisco IOS Gateways to Enable Fax Pass-throughRecommended Setting This setting translates to the bitmapRun the following command Enable Fax Pass-through ModePerform the command Disable Fax Relay FeatureConfiguring the Cisco ATA for Fax Mode Using FAX ModeCommon Problems When Using IOS Gateways Debugging the Cisco ATA 186/188 Fax ServicesConfiguring the Cisco ATA for Fax Mode on a Per-Call Basis Configuring the Cisco IOS Gateway for Fax ModePort Problem ActionCisco ATA, and 0x0012XXXX for the Phone 2 port For fax pass-through mode, AudioMode should be set toPrserv Overview, Analyzing prserv Output for Fax Sessions, Using prserv for Diagnosing Fax ProblemsPrserv Overview Analyzing prserv Output for Fax SessionsTerminating-Gateway Example Log event DescriptionPossible Reasons for Failure Originating-Gateway ExampleTo use rtpcatch, follow these steps Using rtpcatch for Diagnosing Fax ProblemsRtpcatch Overview Output Files Example of rtpcatchExplanation CED tone Detected Both sides use G.711 for the entire fax session Fax relay mode Cisco fax relay modeAnalyzing rtpcatch Output for Fax Sessions AnalysisExample 7-4 T38 Fax Relay Mode Possible Cause for Failure Using rtpcatch to Analyze Common Causes of FailureCisco fax relay option is not disabled on the gateway Possible Causes for FailureExample 7-9 Fax Pass-through Mode Failure Rtpcatch Limitations Definitions Upgrading the Signaling Image from a Tftp ServerSyntax of upgradecode Parameter Preliminary Steps, Running the Executable File, Upgrading the Signaling Image ManuallyUpgradecode parameter value could be ProcessRunning the Executable File Upgrade RequirementsPreliminary Steps Syntax Upgrade Procedure and VerificationTo perform the upgrade, follow these steps Procedure to Upgrade Signaling ImageUsing a Web Browser, Using the Voice Configuration Menu, Confirming a Successful Signaling Image UpgradeUsing a Web Browser Using the Voice Configuration Menu General Troubleshooting Tips TroubleshootingSymptoms and Actions Installation and Upgrade Issues Debugging Ring Load per RJ-11 FXS Port Maximum Distance Frequently Asked QuestionsFeet 975 m Feet 762 mContacting TAC OL-4008-01 Common Supplementary Services Changing Call CommandsCancelling a Supplementary Service Making a Conference Call in the United States Caller IDCall-Waiting Caller ID Calling Line Identification Presentation Making a Conference Call in SwedenCall Waiting in the United States Call Waiting in SwedenCalling Line Identification Restriction in Sweden About Calling Line Identification RestrictionVoice Menu Option Code Description Table B-1lists codes to return basic Cisco ATA informationPassword associated with the secondary phone line Table B-2lists configuration codesPassword associated with the primary phone line UID0 or LoginID0User ID telephone number for the Phone 2 port User ID telephone number for the Phone 1 portOption Code Description Specification Physical SpecificationsThis section describes Cisco ATA specifications Dimensions WeightDescription Specification Electrical SpecificationsEnvironmental Specifications Immunity SpecificationsPhysical Interfaces Ringing CharacteristicsSoftware Specifications Appendix C Cisco ATA Specifications Software Specifications Sccp OL-4008-01 Supported H.323 Messages SignalingStep Action Description Signaling ScenariosEndpoint-to-Gatekeeper Registration Table D-2 Log Listings Table D-2 Log Listings Endpoint-to-Endpoint Call Setup with a Common Gatekeeper Step Table D-4 Log Listings Table D-4 Log Listings Table D-4 Log Listings Table D-4 Log Listings Table D-4 Log Listings Table D-4 Log Listings Table D-4 Log Listings Call Setup from H.323 Network to Circuit Switched Network CSN/PSTN Action DescriptionStep Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Table D-6 Log Listings Null Table D-6 Log Listings GL-1 GL-2 GL-3 GL-4 GL-5 Signaling connection control partMessages can be part of Sgcp and Mgcp messages GL-6 GL-7 Business-class services for Internet telephonyTraffic Allow you to define your own customized markup languageGL-8 IN-1 NumericsIN-2 IN-3 Environmental specifications C-2Ethernet ports Example configuration text file IN-4 Http resetHotline/warmline Http refresh IN-5 PlarIN-6 RJ-45 LEDIN-7 TroubleshootingIN-8
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ATA 188 specifications

The Cisco Systems ATA 188 is a versatile Analog Telephone Adapter designed to facilitate the integration of traditional telephone systems with Voice over Internet Protocol (VoIP) networks. This device has been key in bridging the gap between legacy telephony and modern IP-based communication, allowing users to leverage their existing analog phones while enjoying the benefits of digital connectivity.

One of the main features of the ATA 188 is its ability to connect regular analog phones to a VoIP network, enabling users to make and receive calls over the internet. This significantly reduces calling costs, especially for long-distance and international calls. The ATA 188 supports two phone lines, allowing simultaneous voice calls. This dual-line capability makes it a suitable choice for small businesses or home offices that require multiple lines without the need for extensive infrastructure.

The device is equipped with various technologies that enhance its functionality. It supports the Session Initiation Protocol (SIP) and H.323, making it compatible with a wide range of VoIP service providers. Additionally, the ATA 188 features Quality of Service (QoS) settings, which prioritize voice traffic over the internet, ensuring clear voice quality without interruptions or delays. This is essential for maintaining a professional communication experience, especially in business environments.

Another characteristic of the ATA 188 is its user-friendly configuration interface. It allows users to easily set up and manage their devices through a web-based portal. The configuration process is straightforward, with options to adjust settings such as codec selection, call features including call waiting, and call forwarding functionalities.

Security is also a priority for the ATA 188, as it provides robust protocols to protect call data. The device supports Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS) to encrypt voice traffic and manage signaling securely. This ensures that sensitive conversations remain confidential.

Overall, the Cisco Systems ATA 188 is a reliable and efficient solution for users looking to transition from traditional telephony to VoIP. Its dual-line capacity, compatibility with multiple VoIP standards, user-friendly configuration, and built-in security features make it a valuable asset for both personal and professional communication solutions. In an ever-evolving telecommunications landscape, the ATA 188 remains a relevant and practical choice for integrating legacy telephony with modern internet-based services.