Cisco Systems Comprehensive Guide to Cisco ATA 186 Dial Plan Commands and Parameters

Page 38

Chapter 5 Parameters and Defaults

Optional Feature Parameters

About Dial Plan Commands

Note

Note

Note

The following list contains rules for Cisco ATA dial plans:

. —Wildcard, match any digit entered.

- —Additional digits can be entered. This command can be used only at the end of a dial plan rule (for example, 1408t5- is legal usage of the - command, but 1408t5-3... is illegal).

>#—Defines the # character as a termination character. When the termination character is entered, the dial string is automatically sent. The termination character can be entered only after at least one user-entered digit matches a dial plan rule. Alternatively, the command >* can be used to define * as the termination character.

tn— Defines the timeout value n, in the unit of seconds, for the interdigit timer. Valid values are 0-9 and a-z, where a-z indicates a range of 10 to 36.

rn—Repeat the last pattern n times, where n is 0-9 or a-z. The values a-z indicate a range of 10 to 36. Use the repeat modifier to specify more rules in less space.

The commands ># and tn are modifiers, not patterns, and are ignored by the rn command.

—Used to separate multiple dial plan rules.

^—Logical not. Match any character except the character immediately following the ^ command.

S—Seize rule matching. If a dial plan rule matches the sequence of digits entered by the user to this point, and the modifier S is the next command in the dial plan rule, all other rules are negated for the remainder of the call (for example, a dial plan beginning with *S will be the only one in effect if the user first enters the * key).

All rules apply in the order listed (whichever rule is completely matched first will immediately send the dial string).

No syntax check is performed by the actual implementation. The administrator has the responsibility of making sure that the dial plan is syntactically valid.

Dial Plan Example 1 (Default Dial Plan)

The following dial plan:

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

consists of the following rules:

*St4-—If the first digit entered is *, all other dial plan rules are voided. Additional digits can be entered after the initial * digit, and the timeout before automatic dial string send is four seconds.

#St4—Same as above, except with # as the initial digit entered.

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

1>#t8.r9t2—If the first digit entered is 1, the timeout before automatic send is eight seconds. 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 two seconds. The user can enter more digits until the dial string is sent by the timeout or by the user entering the # character.

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

5-38

OL-4008-01

 

 

Image 38
Contents Parameters and Defaults A P T E RConfiguration Text File Template Description UIPasswordUser Interface UI Parameter Value TypeToConfig Configuration-Complete ParameterParameters for Configuration Method and Encryption UseTFTP SettingsRange Default Voice Configuration Menu Access Code TftpURLUseTFTP, CfgInterval, CfgIntervalEncryptKey UseTFTP, TftpURL,320 Network ParametersDhcp DHCP, StaticIp, StaticRoute, StaticNetMask,StaticRoute Voice Configuration Menu Access Code Related ParametersStaticIp DHCP, StaticRoute, StaticNetMask,DHCP, StaticIp, StaticNetMask, Account Information ParametersStaticNetMask 255.255.255.0UID0 UID1, PWD0, PWD1, UseLoginID, LoginID0, LoginID1,This parameter is the password for the Phone 1 port PWD0UID0, PWD0, PWD1, UseLoginID, LoginID0, LoginID1, UID1Gateway This parameter is the password for the Phone 2 portPWD1 LoginID1, PWD0, PWD1, UseLoginID, AutMethod, UseLoginIDLoginID0 AutMethod LoginID1LoginID0, PWD0, PWD1, UseLoginID, AutMethod, BitmapAltGk, AltGkTimeOut, GkTimeToLive, GkId, LoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP,GkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId, Gatekeeper ParametersAltGk, ConnectMode, page 5-28-Bit AltGkTimeOutAltGk GkId Default RangeGkTimeToLive Operating Parameters Mode ParameterUse H.323 mode -Use SIP mode UseSIPMediaPort DNS2IP, UDPTOS, SigTimer, OpFlags, VLANSetting,UDPTOS, VLANSetting, LBRCodecAudioMode, page 5-20-Bits 1 TxCodec, RxCodec, NumTxFrames, AudioModeRxCodec, TxCodec, LBRCodec, ConnectMode, RxCodec,RxCodec Bit Number DefinitionLBRCodec, NumTxFrames, RxCodec, AudioMode, TxCodecLBRCodec, NumTxFrames, TxCodec, AudioMode, LBRCodec, RxCodec, TxCodec, NumTxFramesExamples Bit Number CallFeaturesCallCmd, 315 PaidFeaturesCallFeatures, CallCmd, CallerIdMethod, SigTimer, Call waitingCallerIdMethod 316 Polarity0x00019e60 ConnectMode Use G.711A-law for fax pass-through codec TimeZoneUse G.711µ-law for fax pass-through codec NTPIP, AltNTPIP, NtpipAltNTPIP AltNTPIP, TimeZone,Udptos DNS1IPDNS2IP NTPIP, TimeZone,SigTimer TftpURL, DHCP, VLANSetting, OpFlagsVLANSetting Syntax Optional Feature ParametersNPrintf 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 PrefixTone Parameter Syntax Call-Progress Tone ParametersList of Call-Progress Tone Parameters This section contains the following topicsUse the following formula to calculate the scaling factor a How to Calculate Scaling FactorsRecommended Values DialTone Default values for the nine-integer arraySpecific Call-Progress Tone Parameter Information 920ReorderTone Cisco ATA plays the busy tone when the callee is busyBusyTone 921923 RingbackToneCallWaitTone 924925 CallCmdAlertTone 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.