Cisco Systems ATA 186 manual Specific Call-Progress Tone Parameter Information, DialTone, 920

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Chapter 5 Parameters and Defaults

Optional Feature Parameters

The following settings are recommended for Sweden:

DialTone = "1,30959,0,4253,0, 1, 0, 0,1500" (approximately -5 dBm)

BusyTone = "1,30959,0,2392,0, 0, 2000, 2000,0" (approximately -10 dBm)

ReorderTone = "1,30959,0,2392,0, 0, 2000, 6000,0" (approximately -10 dBm)

RingBackTone = "1,30959,0,2392,0, 0, 8000, 40000,0" (approximately -10 dBm)

CallWaitTone = "1,30959,0,2392,0, 0, 1600, 4000,11200" (approximately -10 dBm)

AlertTone = "1,30959,0,2392,0, 0, 480, 480,1920" (approximately -10 dBm)

Specific Call-Progress Tone Parameter Information

Brief descriptions, and lists of default values and the voice configuration menu code for each Cisco ATA tone parameter, are described in the following sections:

DialTone, page 5-43

BusyTone, page 5-44

ReorderTone, page 5-44

RingbackTone, page 5-45

CallWaitTone, page 5-45

AlertTone, page 5-46

DialTone

Description

The Cisco ATA plays the dial tone when it is ready to accept the first digit of a remote address to make an outgoing call.

Default values for the nine-integer array

ntone—2

freq0—31538

freq1—30831

level0—1380

level1—1740

steady—1

on-time—0

off-time—0

total time to play tone—1000

Voice Configuration Menu Access Code

920

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

 

OL-4008-01

5-43

 

 

 

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Contents A P T E R Parameters and DefaultsConfiguration Text File Template Value Type UIPasswordUser Interface UI Parameter DescriptionParameters for Configuration Method and Encryption Configuration-Complete ParameterToConfig TftpURL SettingsRange Default Voice Configuration Menu Access Code UseTFTPUseTFTP, TftpURL, CfgIntervalEncryptKey UseTFTP, CfgInterval,DHCP, StaticIp, StaticRoute, StaticNetMask, Network ParametersDhcp 320DHCP, StaticRoute, StaticNetMask, Voice Configuration Menu Access Code Related ParametersStaticIp StaticRoute255.255.255.0 Account Information ParametersStaticNetMask DHCP, StaticIp, StaticNetMask,PWD0 UID1, PWD0, PWD1, UseLoginID, LoginID0, LoginID1,This parameter is the password for the Phone 1 port UID0UID1 UID0, PWD0, PWD1, UseLoginID, LoginID0, LoginID1,PWD1 This parameter is the password for the Phone 2 portGateway LoginID0 UseLoginIDLoginID1, PWD0, PWD1, UseLoginID, AutMethod, Bitmap LoginID1LoginID0, PWD0, PWD1, UseLoginID, AutMethod, AutMethodGatekeeper Parameters LoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP,GkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId, AltGk, AltGkTimeOut, GkTimeToLive, GkId,AltGk AltGkTimeOutAltGk, ConnectMode, page 5-28-Bit GkTimeToLive Default RangeGkId UseSIP 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, TxCodec, AudioMode, TxCodecLBRCodec, NumTxFrames, RxCodec, AudioMode, Examples NumTxFramesLBRCodec, RxCodec, TxCodec, CallCmd, CallFeaturesBit Number Call waiting PaidFeaturesCallFeatures, CallCmd, CallerIdMethod, SigTimer, 315CallerIdMethod 0x00019e60 Polarity316 ConnectMode Use G.711µ-law for fax pass-through codec TimeZoneUse G.711A-law for fax pass-through codec AltNTPIP, TimeZone, NtpipAltNTPIP NTPIP, AltNTPIP,NTPIP, TimeZone, DNS1IPDNS2IP UdptosSigTimer OpFlags TftpURL, DHCP, VLANSetting,VLANSetting Example Optional Feature ParametersNPrintf SyntaxRingOnOffTime Default Recommended ValuesIPDialPlan Additional DialPlan Information DialPlanDial Plan Example 1 Default Dial Plan About Dial Plan CommandsFollowing dial plan Dial Plan Example Dial Plan Blocking In RuleFollowing dial plans Rule to Support Dial Prefix Rule to Support Hotline/WarmlineThis section contains the following topics Call-Progress Tone ParametersList of Call-Progress Tone Parameters Tone Parameter SyntaxRecommended Values How to Calculate Scaling FactorsUse the following formula to calculate the scaling factor a 920 Default values for the nine-integer arraySpecific Call-Progress Tone Parameter Information DialTone921 Cisco ATA plays the busy tone when the callee is busyBusyTone ReorderTone924 RingbackToneCallWaitTone 923Maximum of 248 characters CallCmdAlertTone 925930 CallFeatures, PaidFeatures, CallerIdMethod, SigTimer,OL-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.