Cisco Systems ATA 186 manual OpFlags, TftpURL, DHCP, VLANSetting

Page 33

Chapter 5 Parameters and Defaults

Operating Parameters

Table 5-5

SigTimer Parameter Bit Definitions (continued)

 

 

 

Bit Number

 

Definition

 

 

 

20-25

 

Not used by H.323.

 

 

 

26-27

 

Minimum hook flash time—The minimum on-hook time required for hook flash event.

 

 

Range: 0 to 3

 

 

Default: 0 (60 ms)

 

 

Other possible values: 1=100 ms, 2=200 ms, 3=300 ms.

 

 

 

28-31

 

Maximum hook flash time—The maximum on-hook time allowed for hook flash event.

 

 

Range: 0 to 15

 

 

Default: 0 (1000 ms)

 

 

Other possible values: 1=100 ms, 2=200 ms, 3=300 ms, 4=400 ms, 5=500 ms, 6=600 ms, 7=700 ms, 8=800

 

 

ms, 9=900 ms, 10=1000 ms, 11=1100 ms, 12=1200 ms, 13=1300 ms, 14=1400 ms, 15=1500 ms.

 

 

 

OpFlags

Description

This parameter enables/disables various operational features.

See Table 5-6 on page 5-33for bit definitions of this parameter.

Value Type

Bitmap

Default 0x2

Voice Configuration Menu Access Code 323

Related Parameters

•TftpURL, page 5-5

•DHCP, page 5-7

•VLANSetting, page 5-34

Table 5-6 OpFlags Parameter Operational Features to Turn On or Off

Bit Number Definition

0If Bit 0 = 0, the TFTP configuration filename supplied by the DHCP server overwrites the default filename for each Cisco ATA.

If Bit 0 = 1, the default Cisco ATA filename is always used.

1If Bit 1 = 0, the Cisco ATA probes the static network router during the power-up process. If Bit 1 = 1, static network router probing is disabled.

2Reserved.

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

 

OL-4008-01

5-33

 

 

 

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Contents A P T E R Parameters and DefaultsConfiguration Text File Template User Interface UI Parameter UIPasswordDescription Value TypeConfiguration-Complete Parameter Parameters for Configuration Method and EncryptionToConfig 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,This parameter is the password for the Phone 2 port PWD1Gateway UseLoginID LoginID0LoginID1, PWD0, PWD1, UseLoginID, AutMethod, LoginID0, PWD0, PWD1, UseLoginID, AutMethod, LoginID1AutMethod BitmapGkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId, LoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP,AltGk, AltGkTimeOut, GkTimeToLive, GkId, Gatekeeper ParametersAltGkTimeOut AltGkAltGk, ConnectMode, page 5-28-Bit Default Range GkTimeToLiveGkId 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 RxCodecTxCodec LBRCodec, NumTxFrames, TxCodec, AudioMode,LBRCodec, NumTxFrames, RxCodec, AudioMode, NumTxFrames ExamplesLBRCodec, RxCodec, TxCodec, CallFeatures CallCmd,Bit Number CallFeatures, CallCmd, CallerIdMethod, SigTimer, PaidFeatures315 Call waitingCallerIdMethod Polarity 0x00019e60316 ConnectMode TimeZone Use G.711µ-law for fax pass-through codecUse G.711A-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 ExampleDefault Recommended Values RingOnOffTimeIPDialPlan Additional DialPlan Information DialPlanAbout 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 Dial Prefix Rule to Support Hotline/WarmlineList of Call-Progress Tone Parameters 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 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.
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