Cisco Systems ATA 186 manual Dial Plan Blocking In Rule, Dial Plan Example, Following dial plans

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Chapter 5 Parameters and Defaults

Optional Feature Parameters

0>#t811.rat4—If the first digit entered is 0, the timeout before automatic send is eight seconds, and the terminating character # can be entered at any time to manually send the dial string. If the first three digits entered are 011, then, after an additional 11 digits are entered, the timeout before an automatic send changes to four seconds. The user can enter more digits until the dial string is sent by the timeout or by the user entering the # character.

^1t4>#.—If the first digit entered is anything other than 1, the timeout before an automatic send is four seconds. The terminating character # can be entered at any time to manually send the dial string. The user can enter more digits until the dial string is sent by the timeout or by the user entering the

# character.

Dial Plan Example 2

The following dial plans:

.t7>#

t4-9111t7>#

..........t1-0t4>#.t7-

or

.t7>#r6t4-9111t7>#.r9t1-0t4>#.t7-

consist of the following rules:

.t7>#r6t4-—You must enter at least one digit. After the first digit is entered and matched by the dial plan, the timeout before an automatic send is seven seconds, and the terminating character # can be entered at any time to manually send the dial string. After seven digits are entered, the timeout before an automatic send changes to two seconds. The - symbol at the end of the rule allows further digits to be entered until the dial string is sent by the timeout or the user entering the # character.

911—If the dial string 911 is entered, send this string immediately.

1t7>#.r9t1—If the first digit entered is 1, the timeout before an automatic send is seven seconds, and the terminating character # can be entered at any time to manually send the dial string. After the 11th digit is entered, the timeout before an automatic send changes to one second. The user can enter more digits until the dial string is sent by the timeout or by the user entering the # character.

0t4>#.t7—If the first digit entered is 0, the timeout before an automatic send is four seconds, and the terminating character # can be entered at any time to manually send the dial string. After the second digit is entered, the timeout before an automatic send changes to seven seconds. The user can enter more digits until the dial string is sent by the timeout or by the user entering the # character.

Dial Plan Blocking (In Rule)

Dial plan blocking can be used to reduce the occurrences of invalid dialed digits being sent and can prevent the dialed string of a specified pattern from being sent. By adding dial plan blocking, dialed digits are discarded after the interdigit timer expires unless one of the specified matching rules is met.

In addition, the default nine-second global interdigit timeout value is also modified with the value specified in the dial plan blocking command:

In

where n specifies the global interdigit timeout and the valid values are 1-9 and a-z (10-35).

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

 

OL-4008-01

5-39

 

 

 

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Contents A P T E R Parameters and DefaultsConfiguration Text File Template Value Type UIPasswordUser Interface UI Parameter DescriptionConfiguration-Complete Parameter Parameters for Configuration Method and EncryptionToConfig TftpURL SettingsRange Default Voice Configuration Menu Access Code UseTFTPUseTFTP, TftpURL, CfgIntervalEncryptKey UseTFTP, CfgInterval,DHCP, StaticIp, StaticRoute, StaticNetMask, Network ParametersDhcp 320DHCP, StaticRoute, StaticNetMask, Voice Configuration Menu Access Code Related ParametersStaticIp StaticRoute255.255.255.0 Account Information ParametersStaticNetMask DHCP, StaticIp, StaticNetMask,PWD0 UID1, PWD0, PWD1, UseLoginID, LoginID0, LoginID1,This parameter is the password for the Phone 1 port UID0UID1 UID0, PWD0, PWD1, UseLoginID, LoginID0, LoginID1,This parameter is the password for the Phone 2 port PWD1Gateway UseLoginID LoginID0LoginID1, PWD0, PWD1, UseLoginID, AutMethod, Bitmap LoginID1LoginID0, PWD0, PWD1, UseLoginID, AutMethod, AutMethodGatekeeper Parameters LoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP,GkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId, AltGk, AltGkTimeOut, GkTimeToLive, GkId,AltGkTimeOut AltGkAltGk, ConnectMode, page 5-28-Bit Default Range GkTimeToLiveGkId UseSIP Mode ParameterUse H.323 mode -Use SIP mode Operating ParametersLBRCodec DNS2IP, UDPTOS, SigTimer, OpFlags, VLANSetting,UDPTOS, VLANSetting, MediaPortLBRCodec, ConnectMode, RxCodec, AudioModeRxCodec, TxCodec, AudioMode, page 5-20-Bits 1 TxCodec, RxCodec, NumTxFrames,Bit Number Definition RxCodecTxCodec LBRCodec, NumTxFrames, TxCodec, AudioMode,LBRCodec, NumTxFrames, RxCodec, AudioMode, NumTxFrames ExamplesLBRCodec, RxCodec, TxCodec, CallFeatures CallCmd,Bit Number Call waiting PaidFeaturesCallFeatures, CallCmd, CallerIdMethod, SigTimer, 315CallerIdMethod Polarity 0x00019e60316 ConnectMode TimeZone Use G.711µ-law for fax pass-through codecUse G.711A-law for fax pass-through codec AltNTPIP, TimeZone, NtpipAltNTPIP NTPIP, AltNTPIP,NTPIP, TimeZone, DNS1IPDNS2IP UdptosSigTimer OpFlags TftpURL, DHCP, VLANSetting,VLANSetting Example Optional Feature ParametersNPrintf SyntaxDefault 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/WarmlineThis section contains the following topics Call-Progress Tone ParametersList of Call-Progress Tone Parameters Tone Parameter SyntaxHow to Calculate Scaling Factors Recommended ValuesUse the following formula to calculate the scaling factor a 920 Default values for the nine-integer arraySpecific Call-Progress Tone Parameter Information DialTone921 Cisco ATA plays the busy tone when the callee is busyBusyTone ReorderTone924 RingbackToneCallWaitTone 923Maximum of 248 characters CallCmdAlertTone 925930 CallFeatures, PaidFeatures, CallerIdMethod, SigTimer,OL-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.