Cisco Systems ATA 186 manual TimeZone, Use G.711µ-law for fax pass-through codec

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Chapter 5 Parameters and Defaults

Operating Parameters

Table 5-4 ConnectMode Parameter Bit Definitions (continued)

Bit Number

Definition

 

 

13

0—Use G.711µ-law for fax pass-through codec.

 

1—Use G.711A-law for fax pass-through codec.

 

 

14-15

0—Use fax pass-through.

 

1—Use codec negotiation in sending fax.

 

2—Reserved.

 

3—Reserved.

 

 

16—18

Not used for H.323.

 

 

19

0—Disable sending ringback tone to the caller.

 

1—Enable sending ringback tone to the caller.

 

 

20-22

Not used by H.323.

 

 

23

0—Do not allow the end-user to configure the permanent default call-waiting setting for every call.

 

Instead, use the service provider's default call-waiting setting for every call.

 

1—Allows the end-user to configure the permanent default call-waiting setting for every call, and

 

override the service provider's default call-waiting setting.

 

 

24

0/1—Disable/enable the mixing of audio and call waiting tone during a call.

 

 

25

Not used for H.323.

 

 

26 to 31

Reserved.

 

 

TimeZone

Description

This parameter is the timezone offset from Greenwich Mean Time (GMT) for time-stamping incoming calls with local time (to use for Caller ID display, for example).

Local time is generated by the following formula:

Local Time=GMT + TimeZone, if TimeZone <= 12

Local Time=GMT + TimeZone - 25, if TimeZone > 12

Value Type

Integer

Range 0-24

Default 17

Voice Configuration Menu Access Code 302

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

 

OL-4008-01

5-29

 

 

 

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Contents A P T E R Parameters and DefaultsConfiguration Text File Template User Interface UI Parameter UIPasswordDescription Value TypeToConfig Configuration-Complete ParameterParameters for Configuration Method and Encryption Range Default Voice Configuration Menu Access Code SettingsUseTFTP TftpURLEncryptKey CfgIntervalUseTFTP, CfgInterval, UseTFTP, TftpURL,Dhcp Network Parameters320 DHCP, StaticIp, StaticRoute, StaticNetMask,StaticIp Voice Configuration Menu Access Code Related ParametersStaticRoute DHCP, StaticRoute, StaticNetMask,StaticNetMask Account Information ParametersDHCP, StaticIp, StaticNetMask, 255.255.255.0This parameter is the password for the Phone 1 port UID1, PWD0, PWD1, UseLoginID, LoginID0, LoginID1,UID0 PWD0UID1 UID0, PWD0, PWD1, UseLoginID, LoginID0, LoginID1,Gateway This parameter is the password for the Phone 2 portPWD1 LoginID1, PWD0, PWD1, UseLoginID, AutMethod, UseLoginIDLoginID0 LoginID0, PWD0, PWD1, UseLoginID, AutMethod, LoginID1AutMethod BitmapGkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId, LoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP,AltGk, AltGkTimeOut, GkTimeToLive, GkId, Gatekeeper ParametersAltGk, ConnectMode, page 5-28-Bit AltGkTimeOutAltGk GkId Default RangeGkTimeToLive Use H.323 mode -Use SIP mode Mode ParameterOperating Parameters UseSIPUDPTOS, VLANSetting, DNS2IP, UDPTOS, SigTimer, OpFlags, VLANSetting,MediaPort LBRCodecRxCodec, TxCodec, AudioModeAudioMode, page 5-20-Bits 1 TxCodec, RxCodec, NumTxFrames, LBRCodec, ConnectMode, RxCodec,Bit Number Definition RxCodecLBRCodec, NumTxFrames, RxCodec, AudioMode, TxCodecLBRCodec, NumTxFrames, TxCodec, AudioMode, LBRCodec, RxCodec, TxCodec, NumTxFramesExamples Bit Number CallFeaturesCallCmd, CallFeatures, CallCmd, CallerIdMethod, SigTimer, PaidFeatures315 Call waitingCallerIdMethod 316 Polarity0x00019e60 ConnectMode Use G.711A-law for fax pass-through codec TimeZoneUse G.711µ-law for fax pass-through codec AltNTPIP NtpipNTPIP, AltNTPIP, AltNTPIP, TimeZone,DNS2IP DNS1IPUdptos NTPIP, TimeZone,SigTimer OpFlags TftpURL, DHCP, VLANSetting,VLANSetting NPrintf Optional Feature ParametersSyntax ExampleIPDialPlan Default Recommended ValuesRingOnOffTime Additional DialPlan Information DialPlanFollowing dial plan About Dial Plan CommandsDial Plan Example 1 Default Dial Plan Following dial plans Dial Plan Blocking In RuleDial Plan Example Rule to Support Dial Prefix Rule to Support Hotline/WarmlineList of Call-Progress Tone Parameters Call-Progress Tone ParametersTone Parameter Syntax This section contains the following topicsUse the following formula to calculate the scaling factor a How to Calculate Scaling FactorsRecommended Values Specific Call-Progress Tone Parameter Information Default values for the nine-integer arrayDialTone 920BusyTone Cisco ATA plays the busy tone when the callee is busyReorderTone 921CallWaitTone RingbackTone923 924AlertTone CallCmd925 Maximum of 248 characters930 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.