Cisco Systems ATA 186 manual CallFeatures, CallCmd, Bit Number

Page 24

Chapter 5 Parameters and Defaults

Operating Parameters

CallFeatures

 

 

 

 

Description

 

 

 

 

Disable/enable CallFeatures by setting each corresponding bit to 0 or 1.

 

 

 

 

The lower 16 bits are for the Phone 1 port, and the upper 16 bits are for the Phone 2 port. Table 5-2

 

 

 

 

provides definitions of each bit.

 

 

 

 

 

 

 

Note

The subscribed features that can be permanently disabled by the user are CLIP_CLIR, call waiting and

 

 

 

 

Fax mode. A subscribed service enable/disabled by the user can be disabled/enabled dynamically on a

 

 

 

 

per-call basis.

 

 

 

 

 

 

 

 

 

 

Value Type

 

 

 

 

Bitmap

 

 

 

 

Default

 

 

 

 

0xffffffff

 

 

 

 

Voice Configuration Menu Access Code

 

 

 

 

314

 

 

 

 

 

 

Related Parameters

 

 

 

 

PaidFeatures, page 5-25

 

 

 

 

CallCmd, page 5-46

 

 

 

 

CallerIdMethod, page 5-26

Table 5-2 CallFeatures Parameter Bit Definitions

 

 

 

 

 

 

 

Bit

 

 

 

 

 

 

Number

Definition

 

 

 

 

 

 

0 and 16

Not used for H.323.

 

 

 

1 and 17

Not used for H.323.

 

 

 

2 and 18

Not used for H.323.

 

 

 

3 and 19

CLIP (if set to 1)/CLIR (if set to 0)

 

 

 

 

4 and 20

Call waiting

 

 

 

 

 

5 and 21

Three-way calling

 

 

 

6 and 22

Not used for H.323.

 

 

 

7 and 23

Not used for H.323.

 

 

 

 

 

8 and 24

Caller ID.

 

 

 

 

 

 

9 and 25

Not used for H.323.

 

 

 

10 and 26

Not used for H.323.

 

 

 

11 and 27

Call Waiting Caller ID.

 

 

 

 

 

 

 

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

5-24

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.