Cisco Systems Official Manual for the Cisco ATA 186: Configuration Parameters and Settings

Page 36

Chapter 5 Parameters and Defaults

Optional Feature Parameters

Voice Configuration Menu Access Code

81

RingOnOffTime

Description

This parameter specifies the ringer cadence pattern, expressed as a triplet of integers “a,b, and c”.

a—Number of seconds to turn the ring ON.

b—Number of seconds to turn the ring OFF.

c—The ring frequency, fixed at 25.

Value Type

List of three integer values, separated by commas

Range

1-65535

Default

2, 4, 25

Recommended Values:

United States —2,4,25

Sweden — 1,5,25

Voice Configuration Menu Access Code

929

IPDialPlan

Description

This parameter allows for detection of IP-like destination address in DialPlan. Three values are valid:

0—String is dialed as is and not treated as an IP address.

1—When the Cisco ATA detects two asterisks (**), IPDialPlan takes over. The user enters the pound

(#)key to terminate the digit collection, and the interdigit timeout default is not used.

2—When IPDialPlan is set to 2, three asterisks (***) are required for IPDialPlan to take effect. All other values are currently undefined.

Value Type

Integer

Range

0, 1 or 2

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

5-36

OL-4008-01

 

 

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Contents Parameters and Defaults A P T E RConfiguration Text File Template UIPassword User Interface UI ParameterDescription Value TypeConfiguration-Complete Parameter Parameters for Configuration Method and EncryptionToConfig Settings Range Default Voice Configuration Menu Access CodeUseTFTP TftpURLCfgInterval EncryptKeyUseTFTP, CfgInterval, UseTFTP, TftpURL,Network Parameters Dhcp320 DHCP, StaticIp, StaticRoute, StaticNetMask,Voice Configuration Menu Access Code Related Parameters StaticIpStaticRoute DHCP, StaticRoute, StaticNetMask,Account Information Parameters StaticNetMaskDHCP, StaticIp, StaticNetMask, 255.255.255.0UID1, PWD0, PWD1, UseLoginID, LoginID0, LoginID1, This parameter is the password for the Phone 1 portUID0 PWD0UID0, PWD0, PWD1, UseLoginID, LoginID0, LoginID1, UID1This parameter is the password for the Phone 2 port PWD1Gateway UseLoginID LoginID0LoginID1, PWD0, PWD1, UseLoginID, AutMethod, LoginID1 LoginID0, PWD0, PWD1, UseLoginID, AutMethod,AutMethod BitmapLoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP, GkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId,AltGk, AltGkTimeOut, GkTimeToLive, GkId, Gatekeeper ParametersAltGkTimeOut AltGkAltGk, ConnectMode, page 5-28-Bit Default Range GkTimeToLiveGkId Mode Parameter Use H.323 mode -Use SIP modeOperating Parameters UseSIPDNS2IP, UDPTOS, SigTimer, OpFlags, VLANSetting, UDPTOS, VLANSetting,MediaPort LBRCodecAudioMode RxCodec, TxCodec,AudioMode, page 5-20-Bits 1 TxCodec, RxCodec, NumTxFrames, LBRCodec, ConnectMode, RxCodec,RxCodec Bit Number DefinitionTxCodec LBRCodec, NumTxFrames, TxCodec, AudioMode,LBRCodec, NumTxFrames, RxCodec, AudioMode, NumTxFrames ExamplesLBRCodec, RxCodec, TxCodec, CallFeatures CallCmd,Bit Number PaidFeatures CallFeatures, CallCmd, CallerIdMethod, SigTimer,315 Call waitingCallerIdMethod Polarity 0x00019e60316 ConnectMode TimeZone Use G.711µ-law for fax pass-through codecUse G.711A-law for fax pass-through codec Ntpip AltNTPIPNTPIP, AltNTPIP, AltNTPIP, TimeZone,DNS1IP DNS2IPUdptos NTPIP, TimeZone,SigTimer TftpURL, DHCP, VLANSetting, OpFlagsVLANSetting Optional Feature Parameters NPrintfSyntax ExampleDefault Recommended Values RingOnOffTimeIPDialPlan DialPlan Additional DialPlan InformationAbout Dial Plan Commands Dial Plan Example 1 Default Dial PlanFollowing dial plan Dial Plan Blocking In Rule Dial Plan ExampleFollowing dial plans Rule to Support Hotline/Warmline Rule to Support Dial PrefixCall-Progress Tone Parameters List of Call-Progress Tone ParametersTone Parameter Syntax This section contains the following topicsHow to Calculate Scaling Factors Recommended ValuesUse the following formula to calculate the scaling factor a Default values for the nine-integer array Specific Call-Progress Tone Parameter InformationDialTone 920Cisco ATA plays the busy tone when the callee is busy BusyToneReorderTone 921RingbackTone CallWaitTone923 924CallCmd AlertTone925 Maximum of 248 charactersCallFeatures, PaidFeatures, CallerIdMethod, SigTimer, 930OL-4008-01
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ATA 186 specifications

The Cisco Systems ATA 186, or Analog Telephone Adapter, revolutionized the way traditional telephony interacted with Voice over Internet Protocol (VoIP) systems. Designed primarily for home and small office use, the ATA 186 allows users to connect standard analog phones and fax machines to a network, enabling them to take advantage of the benefits of VoIP technology.

One of the defining features of the ATA 186 is its dual port architecture. It includes two FXS ports, allowing users to connect up to two analog telephones. This functionality means that multiple devices can leverage VoIP services simultaneously without the need for separate adapters for each phone. The flexibility of the ATA 186 helps streamline the user experience, facilitating voice communication over an IP network while ensuring users can still use their existing phone equipment.

The ATA 186 employs various technologies to maintain high-quality voice calls. It supports standard voice codecs such as G.711 and G.729, which ensure efficient bandwidth usage while preserving call clarity. The adaptive jitter buffer technology further enhances call quality, compensating for network variations and minimizing latency, which is crucial for clear and uninterrupted conversations.

Additionally, the ATA 186 provides users with advanced calling features that were traditionally available only on PBX systems. These features include caller ID, call waiting, and voicemail functionality, integrating seamlessly with typical telephony services. The device also supports T.38 fax relay, allowing users to send and receive faxes over the internet, thus addressing the needs of environments where fax communication remains essential.

The security of VoIP conversations is also a priority for the ATA 186. It employs encryption protocols such as Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS), ensuring that voice data is securely transmitted across the network. This focus on security helps protect sensitive information transmitted during calls.

Installation and configuration of the ATA 186 is streamlined, with an intuitive web-based interface that simplifies the setup process. This accessibility makes it suitable for users with varying levels of technical expertise, as frequently required adjustments, such as network configurations and firmware updates, can be easily managed.

In conclusion, the Cisco Systems ATA 186 stands out as a versatile and robust solution for users looking to integrate analog phones into a VoIP environment. With its dual port capabilities, high-quality voice codecs, advanced call features, and security measures, it offers a compelling choice for both residential and commercial users seeking seamless telephony integration. As technology evolves, devices like the ATA 186 remain cornerstones in bridging traditional telephony with modern communication systems.