Cisco Systems ATA 186 manual ConnectMode

Page 28

Chapter 5 Parameters and Defaults

Operating Parameters

Default

0x00000000

Voice Configuration Menu Access Code

304

ConnectMode

Description

This parameter is a 32-bit bitmap used to control the connection mode of the selected call signaling protocol. Table 5-4 on page 5-28provides bit definitions for this parameter.

Value Type

Bitmap

Default 0x00060400

 

Voice Configuration Menu Access Code

 

311

Table 5-4 ConnectMode Parameter Bit Definitions

 

 

Bit Number

Definition

 

 

0

0—Use slow-start procedure (for H.225/Q.931 and H.245).

 

1—Use fast-start procedure (for H.225/Q.931).

 

 

1

0/1—Disable/enable h245 tunneling.

 

 

2

0—Use the dynamic payload type 126/127 as the RTP payload type (fax pass-through mode) for G.711

 

µ-law/G.711 A-law.

 

1—Use the standard payload type 0/8 as the RTP payload type (fax pass-through mode) for G.711

 

µ-law/G.711 A-law.

 

 

3

0—Do not perform full gatekeeper registration when the Cisco ATA switches to an alternate H.323

 

gatekeeper.

 

1—Perform full gatekeeper registration when the Cisco ATA switches to an alternate H.323 gatekeeper.

 

 

4

0—Denotes a non-Cisco CallManager environment.

 

1—Enable the Cisco ATA to operate in a Cisco CallManager environment.

 

 

5

0/1—Enable/disable two-way cut-through of voice path before the Cisco ATA receives the CONNECT

 

message.

 

 

6

0/1—Disable/enable using the Progress Indicator to determine if ringback is supplied by the far end with

 

RTP.

 

 

7

0/1—Disable/enable fax pass-through redundancy.

 

 

8-12

Specifies the fax pass-through NSE payload type. The value is the offset to the NSE payload base number

 

of 96. The valid range is 0-23; the default is 4.

 

For example, if the offset is 4, the NSE payload type is 100.

 

 

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

5-28

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 TypeParameters for Configuration Method and Encryption Configuration-Complete ParameterToConfig 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, UID1PWD1 This parameter is the password for the Phone 2 portGateway LoginID0 UseLoginIDLoginID1, 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 ParametersAltGk AltGkTimeOutAltGk, ConnectMode, page 5-28-Bit GkTimeToLive Default RangeGkId 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, TxCodec, AudioMode, TxCodecLBRCodec, NumTxFrames, RxCodec, AudioMode, Examples NumTxFramesLBRCodec, RxCodec, TxCodec, CallCmd, CallFeaturesBit Number PaidFeatures CallFeatures, CallCmd, CallerIdMethod, SigTimer,315 Call waitingCallerIdMethod 0x00019e60 Polarity316 ConnectMode Use G.711µ-law for fax pass-through codec TimeZoneUse 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 ExampleRingOnOffTime Default Recommended ValuesIPDialPlan DialPlan Additional DialPlan InformationDial 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 Hotline/Warmline Rule to Support Dial PrefixCall-Progress Tone Parameters List of Call-Progress Tone ParametersTone Parameter Syntax This section contains the following topicsRecommended Values How to Calculate Scaling FactorsUse 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.