Cisco Systems ATA 186 Mode Parameter, Operating Parameters, UseSIP, Use H.323 mode -Use SIP mode

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Chapter 5 Parameters and Defaults

H.323 Mode Parameter

H.323 Mode Parameter

This section describes the UseSIP parameter, which is used to set the Cisco ATA to H.323 mode if you are using the H.323 signaling protocol.

UseSIP

Description

0—Use H.323 mode. 1—Use SIP mode.

Value Type

Boolean

Range

0 or 1

Default

0

Voice Configuration Menu Access Code

38

Operating Parameters

The parameters for configuring codecs, fax features and VLAN settings are included in this section:

MediaPort, page 5-19

LBRCodec, page 5-19

AudioMode, page 5-20

RxCodec, page 5-21

TxCodec, page 5-22

NumTxFrames, page 5-23

CallFeatures, page 5-24

PaidFeatures, page 5-25

CallerIdMethod, page 5-26

Polarity, page 5-27

ConnectMode, page 5-28

TimeZone, page 5-29

NTPIP, page 5-30

AltNTPIP, page 5-30

DNS1IP, page 5-31

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

5-18

OL-4008-01

 

 

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Contents Parameters and Defaults A P T E RConfiguration Text File Template Description UIPasswordUser Interface UI Parameter Value TypeConfiguration-Complete Parameter Parameters for Configuration Method and EncryptionToConfig 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, UID1This parameter is the password for the Phone 2 port PWD1Gateway UseLoginID LoginID0LoginID1, PWD0, PWD1, UseLoginID, AutMethod, AutMethod LoginID1LoginID0, PWD0, PWD1, UseLoginID, AutMethod, BitmapAltGk, AltGkTimeOut, GkTimeToLive, GkId, LoginID0, LoginID1, PWD0, PWD1, NTPIP, AltNTPIP,GkOrProxy, AltGk, AltGkTimeOut, GkTimeToLive, GkId, Gatekeeper ParametersAltGkTimeOut AltGkAltGk, ConnectMode, page 5-28-Bit Default Range GkTimeToLiveGkId 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 DefinitionTxCodec LBRCodec, NumTxFrames, TxCodec, AudioMode,LBRCodec, NumTxFrames, RxCodec, AudioMode, NumTxFrames ExamplesLBRCodec, RxCodec, TxCodec, CallFeatures CallCmd,Bit Number 315 PaidFeaturesCallFeatures, CallCmd, CallerIdMethod, SigTimer, Call waitingCallerIdMethod Polarity 0x00019e60316 ConnectMode TimeZone Use G.711µ-law for fax pass-through codecUse G.711A-law for fax pass-through codec NTPIP, AltNTPIP, NtpipAltNTPIP AltNTPIP, TimeZone,Udptos DNS1IPDNS2IP NTPIP, TimeZone,SigTimer TftpURL, DHCP, VLANSetting, OpFlagsVLANSetting Syntax Optional Feature ParametersNPrintf 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 PrefixTone Parameter Syntax Call-Progress Tone ParametersList of Call-Progress Tone Parameters This section contains the following topicsHow to Calculate Scaling Factors Recommended ValuesUse the following formula to calculate the scaling factor a 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.