Grandstream Networks HT702 Slic Setting, Caller ID Scheme, Polarity Reversal, Loop Current, Gain

Page 43

 

 

 

transmission mode. Select between Disabled, Enabled but not Forced, and Enabled

 

 

 

 

and Forced. Default is Disabled.

 

 

 

 

 

 

 

 

 

 

 

 

SLIC Setting

 

Dependent on standard phone type (and location)

 

 

 

 

 

 

 

 

 

 

 

 

Caller ID Scheme

 

Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, & NTT Japan

 

 

 

 

 

 

 

 

 

 

 

 

Polarity Reversal

 

Default is No. If set to “Yes”, polarity will be reversed upon call establishment and

 

 

 

 

termination.

 

 

 

 

 

 

 

 

 

 

 

 

Loop Current

 

Default is No. Set it to Yes if the traditional PBX you are using with HT70X uses this

 

 

Disconnect

 

method for signaling call termination. Method initiates short voltage drop on the line

 

 

 

 

when remote (VoIP) side disconnects an active call.

 

 

 

 

 

 

 

 

 

 

 

 

Loop Current

 

Default value is 200. Here can be configured duration of such voltage drop described in

 

 

Disconnect Duration

 

topic above. HT70X supports a Duration Range from 100 to 10000 ms.

 

 

 

 

 

 

 

 

 

 

 

 

Enable Hook Flash

 

Default is Yes. If set to “No”, FLASH button could only be used for terminating calls.

 

 

 

 

 

 

 

 

 

 

 

 

Hook Flash Timing

 

Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent

 

 

 

 

unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time

 

 

 

 

value. Default values are 300 minimum and 1100 maximum. HT70X supports a

 

 

 

 

range from 40 to 2000 ms.

 

 

 

 

 

 

 

 

 

 

 

 

On Hook Timing

 

On-hook timing is the minimum time for an on-hook event to be validated. Default

 

 

 

 

value is 400 . HT70X supports a range from 40 to 2000 ms.

 

 

 

 

 

 

 

 

 

 

 

 

Gain

 

Voice path volume adjustment.

 

 

 

 

Rx is a gain level for signals transmitted by FXS

 

 

 

 

Tx is a gain level for signals received by FXS.

 

 

 

 

Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume: -6dB.

 

 

 

 

User can adjust volume of call on either end using the Rx Gain Level parameter and

 

 

 

 

the Tx Gain Level parameter located on the FXS Port Configuration page.

 

 

 

 

If call volume is too low when using the FXS port (ie. the ATA is at user site), adjust

 

 

 

 

volume using the Rx Gain Level parameter under the FXS Port Configuration page.

 

 

 

 

If voice volume is too low at the other end, user may increase the far end volume using

 

 

 

 

the Tx Gain Level parameter under the FXS Port Configuration page.

 

 

 

 

 

 

 

 

 

 

 

 

Disable Line Echo

 

Default is No. If set to “Yes” LEC will be disabled per call base. Recommended for

 

 

Canceller (LEC)

 

FAX/Data calls.

 

 

 

 

 

 

 

 

 

 

 

 

Ring Tones

 

This function lets you configure ring tone cadence preferences. User has 10 choices.

 

 

 

 

The configuration, completed in Distinctive Ring Tones block in the same page, applies

 

 

 

 

to ring tones cadences configured here.

 

 

 

 

 

 

FIRMWARE VERSION 1.0.3.1

HT70X USER MANUAL

Page 43 of 52

Image 43
Contents Grandstream Networks, Inc Index Software Upgrade HT70X Connection Diagram GNU GPL Information Change LOG Warranty Safety CompliancesWelcome Firmware Version Connect Your HT70X Connecting the HT70XEquipment Packaging Link/Activity LED ResetPower LED Internet LEDPhone LED LED-16 LED-13LED-14 LED-15HT70X Featrues Software Features OverviewTelnet Server IP SignalingFeatures Dhcp Server/ClientEMC Hardware SpecificationMain Menu Reset the HT for the new IP address to take EffectBasic Operations Understanding HT70X Voice PromptSee Make a Direct IP Call See Restore Factory Default Setting sectionPhone or Extension Numbers Placing a Phone CallCall Waiting Call HoldCall Transfer Attended Transfer WAY ConferencingFAX Support Enable Srtp Disable Srtp Call FeaturesBlind Transfer Flash/Hook Configuring the HT70X Through Voice Prompts Configuration GuideConfiguring the HT70X VIA WEB Browser Access the WEB Configuration Menu Important SettingsWeb Port End User PasswordDhcp hostname PPPoE passwordPPPoE Service Name IP AddressSet Time Zone Language Reset TypeMTZ+6MDT+5 Allow Dhcp server toFXS Forward Busy DelayedNAT DNDLayer 2 QoS Admin PasswordFirmware Upgrade Layer 3 QoSHTTP/HTTPS Firmware Version ACS URLInternet Telephone Service Provider Primary Radius Auth Syslog LevelSend SIP Log Primary RadiusDownload Device Radius RetryUnregister on Reboot Authenticate IDAuthenticate Password DNS ModeRegistration Retry Wait Enable Ring-TransferDisable Bellcore Style Register ExpirationDelay Proxy-Require SIP T1 TimeoutDisable Dtmf Enable Call FeaturesDisable Reminder Ring Disable Call WaitingDisable Call-Waiting Disable ReceiverUse # as Dial Key Dial Plan PrefixDial Plan Dial Plan Rules Force Timer Special FeatureCaller Request Timer Callee Request TimerVAD Canceller LEC Slic SettingEnable Hook Flash Disable Line EchoHunting Group Enable PortsSIP Use ID Profile IDPort Sip port that will be annexed to the IP address above Saving the Configuration ChangesRebooting the HT70X from Remote Configuration Through a Central ServerFirmware Version Firmware Upgrade Through TFTP/HTTP/HTTPS Software UpgradeInstructions for Upload from Local Directory Instructions for Local Firmware Upgrade Using Tftp ServerFirmware and Configuration File Prefix and Postfix Configuration File DownloadManaging Firmware and Configuration File Download Firmware Version Factory Reset Restore Factory Default SettingReset from web interface Reset Type
Related manuals
Manual 35 pages 43.22 Kb

HT704, HT702 specifications

Grandstream Networks has established itself as a powerful player in the telecommunications industry, especially through its Analog Telephone Adapters (ATAs) like the HT702 and HT704 models. These devices are designed specifically for converting analog voice signals into digital data for seamless integration with Voice over Internet Protocol (VoIP) systems.

The Grandstream HT702 is a two-port ATA that allows users to connect two analog phones to a high-speed internet connection. This model is particularly useful for small businesses or residential users looking to integrate legacy phone systems with modern VoIP technology. One of the key features of the HT702 is its support for the SIP (Session Initiation Protocol) standard, ensuring compatibility with a wide range of VoIP providers. Additionally, it supports advanced telephony features like call transfer, call waiting, and three-way calling, enhancing communication efficiency.

The HT704, on the other hand, is a four-port ATA, offering greater flexibility for users needing to connect multiple devices. It shares many of the same features as the HT702, including SIP support and telephony functionalities, but with additional ports, it is better suited for larger environments. Both models come equipped with advanced security mechanisms, such as AES encryption, which safeguards voice communications.

With user-friendly web-based configuration, the HT702 and HT704 allow for easy setup and management, making them accessible even for those without extensive technical knowledge. Moreover, both devices feature auto-provisioning capabilities, which simplify deployment across multiple units, making them ideal for businesses looking to scale their operations.

The HT702 and HT704 are built with high-quality materials, ensuring durability and long-term performance. They also boast low power consumption, making them an energy-efficient choice. Support for high-definition voice codecs enhances audio quality during calls, providing users with crystal-clear communication.

In summary, Grandstream's HT702 and HT704 Analog Telephone Adapters are robust solutions for anyone looking to transition from traditional telephony to a modern VoIP setup. Their advanced features, security standards, and ease of use make them a reliable choice for both home and business users seeking efficient and effective communication solutions.