Cisco Systems Comprehensive Guide to the Cisco ATA 186 Manual and Configuration Parameters

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Chapter 5 Parameters and Defaults

Operating Parameters

Bit 3: Throughput bit—1=request high throughput

Bit 4: Delay bit—1=request low delay

Bits 5-7: Specify datagram precedence. Values range from 0 (normal precedence) to 7 (network control).

Value Type

Bitmap

Default

0xB8

Voice Configuration Menu Access Code

255

SigTimer

Description

This parameter controls various timeout values. Table 5-5 on page 5-32contains bit definitions of this parameter.

Value Type

Bitmap

Default 0x01418564

 

 

 

 

 

Voice Configuration Menu Access Code

 

 

 

 

318

 

Table 5-5 SigTimer Parameter Bit Definitions

 

 

 

 

 

Bit Number

 

Definition

 

 

 

 

 

0-7

 

 

 

Call waiting period—The period between each burst of call-waiting tone.

 

 

 

 

 

Range: 0 to 255 in 0.1 seconds

 

 

 

 

 

Default: 100 (0x64=100 seconds)

 

 

 

 

 

8-13

 

 

 

Reorder delay—The delay in playing the reorder (fast busy) tone after the far-end caller hangs up.

 

 

 

 

 

Range: 0 to 62 in seconds

 

 

 

 

 

Default—5 (seconds)

 

 

 

 

 

63—Never play the reorder tone.

 

 

 

 

 

14-19

 

 

 

Ring timeout—When a call is not answered, this is the amount of time after which Cisco ATA rejects the

 

 

 

 

 

incoming call.

 

 

 

 

 

Range—0 to 63 in 10 seconds

 

 

 

 

 

Default—6 (60 seconds)

 

 

 

 

 

0—Never times out

 

 

 

 

 

 

 

 

 

 

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

 

 

 

 

 

 

 

 

 

 

 

 

5-32

 

 

 

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 TypeToConfig Configuration-Complete ParameterParameters for Configuration Method and Encryption 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, UID1Gateway This parameter is the password for the Phone 2 portPWD1 LoginID1, PWD0, PWD1, UseLoginID, AutMethod, UseLoginIDLoginID0 LoginID1 LoginID0, PWD0, PWD1, UseLoginID, AutMethod,AutMethod BitmapLoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP, GkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId,AltGk, AltGkTimeOut, GkTimeToLive, GkId, Gatekeeper ParametersAltGk, ConnectMode, page 5-28-Bit AltGkTimeOutAltGk GkId Default RangeGkTimeToLive 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 DefinitionLBRCodec, NumTxFrames, RxCodec, AudioMode, TxCodecLBRCodec, NumTxFrames, TxCodec, AudioMode, LBRCodec, RxCodec, TxCodec, NumTxFramesExamples Bit Number CallFeaturesCallCmd, PaidFeatures CallFeatures, CallCmd, CallerIdMethod, SigTimer,315 Call waitingCallerIdMethod 316 Polarity0x00019e60 ConnectMode Use G.711A-law for fax pass-through codec TimeZoneUse G.711µ-law for fax pass-through codec Ntpip AltNTPIPNTPIP, AltNTPIP, AltNTPIP, TimeZone,DNS1IP DNS2IPUdptos NTPIP, TimeZone,SigTimer TftpURL, DHCP, VLANSetting, OpFlagsVLANSetting Optional Feature Parameters NPrintfSyntax ExampleIPDialPlan Default Recommended ValuesRingOnOffTime DialPlan Additional DialPlan InformationFollowing 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 Hotline/Warmline Rule to Support Dial PrefixCall-Progress Tone Parameters List of 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 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
Related manuals
Manual 208 pages 28.48 Kb Manual 64 pages 26.31 Kb Manual 208 pages 5.39 Kb Manual 166 pages 49.11 Kb Manual 14 pages 13.53 Kb

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.