Cisco Systems ATA 186 manual Additional DialPlan Information

Page 37

Chapter 5 Parameters and Defaults

Optional Feature Parameters

Default

1

Voice Configuration Menu Access Code

310

DialPlan

Description

The programmable dial plan is designed for the service provider to customize the behavior of the Cisco ATA for collecting and sending dialed digits. The dial plan allows the Cisco ATA user to specify the events that trigger the sending of dialed digits. These events include the following:

The termination character has been entered.

The specified dial string pattern has been accumulated.

The specified number of dialed digits has been accumulated.

The specified inter-digit timer has expired.

Value Type

Alphanumeric string

Range

Maximum number of characters is 199.

Default

*St4-#St4-9111>#t8.r9t2-0>#t811.rat4-^1t4>#.-

Voice Configuration Menu Access Code

926

Additional DialPlan Information

The DialPlan section contains the following additional topics that describe commands and rules for creating your own dial plan:

About Dial Plan Commands, page 5-38

Dial Plan Blocking (In Rule), page 5-39

'H' Rule to Support Hotline/Warmline, page 5-40

'P' Rule to Support Dial Prefix, page 5-40

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

 

OL-4008-01

5-37

 

 

 

Image 37
Contents A P T E R Parameters and DefaultsConfiguration Text File Template User Interface UI Parameter UIPasswordDescription Value TypeParameters for Configuration Method and Encryption Configuration-Complete ParameterToConfig 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,PWD1 This parameter is the password for the Phone 2 portGateway LoginID0 UseLoginIDLoginID1, 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 ParametersAltGk AltGkTimeOutAltGk, ConnectMode, page 5-28-Bit GkTimeToLive Default RangeGkId 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, TxCodec, AudioMode, TxCodecLBRCodec, NumTxFrames, RxCodec, AudioMode, Examples NumTxFramesLBRCodec, RxCodec, TxCodec, CallCmd, CallFeaturesBit Number CallFeatures, CallCmd, CallerIdMethod, SigTimer, PaidFeatures315 Call waitingCallerIdMethod 0x00019e60 Polarity316 ConnectMode Use G.711µ-law for fax pass-through codec TimeZoneUse 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 ExampleRingOnOffTime Default Recommended ValuesIPDialPlan Additional DialPlan Information DialPlanDial 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 Dial Prefix Rule to Support Hotline/WarmlineList of Call-Progress Tone Parameters 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 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.