Grandstream Networks Syslog Level, Send SIP Log, Primary Radius Auth, Port HT704 Only

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Syslog Level

 

Select the HT701 to report the log level. Default is NONE. The level is one of EXTRA

 

 

 

 

DEBUG, DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the

 

 

 

 

following events:

 

 

 

 

1.

product model/version on boot up (INFO level)

 

 

 

 

2.

NAT related info (INFO level)

 

 

 

 

3.

sent or received SIP message (DEBUG level)

 

 

 

 

4.

SIP message summary (INFO level)

 

 

 

 

5.

inbound and outbound calls (INFO level)

 

 

 

 

6.

registration status change (INFO level)

 

 

 

 

7.

negotiated codec (INFO level)

 

 

 

 

8.

Ethernet link up (INFO level)

 

 

 

 

9.

SLIC chip exception (WARNING and ERROR levels)

 

 

 

 

10. memory exception (ERROR level)

 

 

 

 

11. Vonage requested syslog style (EXTRA DEBUG level)

 

 

 

 

The Syslog uses USER facility. In addition to standard Syslog payload, it contains the

 

 

 

 

following components:

 

 

 

 

GS_LOG: [device MAC address][error code] error message

 

 

 

 

Example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet link

 

 

 

 

is up

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Send SIP Log

 

Default is No. If Enabled, the HT70X will send a replicate of the SIP packets on the

 

 

 

syslog.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Set the IP or FQDN of the Primary RADIUS Server. HT704 Only

 

 

 

 

Remote Authentication Dial In User Service (RADIUS). HT704 supports RADIUS for

 

 

 

 

authentication, authorization and billing purposes. Primary and secondary RADIUS server

 

 

Primary RADIUS

 

configurations are available to provide redundancy to this feature. In case Primary Radius

 

 

 

server becomes unusable, RADIUS requests will be automatically sent to the secondary

 

 

Server

 

 

 

 

server. When at least one RADUIS server was configured, the device will allow users to

 

 

 

 

 

 

 

 

make phone calls only after authorization from RADIUS server has been received. CDR

 

 

 

 

(Call Detail Record) is also sent to the RADIUS server for billing purposes. RAIDUS

 

 

 

 

server can send requests to terminate calls when run out of pre-paid credit.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Primary RADIUS Auth

 

Default is 1812. Specifies the port to be used for the Primary RADIUS Authentication.

 

 

Port

 

HT704 Only

 

 

 

 

 

 

 

 

 

 

 

 

 

FIRMWARE VERSION 1.0.3.1

HT70X USER MANUAL

Page 34 of 52

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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 Internet LED ResetPower LED Link/Activity LEDPhone LED LED-15 LED-13LED-14 LED-16Software Features Overview HT70X FeatruesDhcp Server/Client IP SignalingFeatures Telnet ServerHardware Specification EMCUnderstanding HT70X Voice Prompt Reset the HT for the new IP address to take EffectBasic Operations Main MenuSee Restore Factory Default Setting section See Make a Direct IP CallPlacing a Phone Call Phone or Extension NumbersCall Waiting Call HoldCall Transfer WAY Conferencing Attended TransferFAX Support Enable Srtp Disable Srtp Call FeaturesBlind Transfer Flash/Hook Configuring the HT70X Through Voice Prompts Configuration GuideConfiguring the HT70X VIA WEB Browser Important Settings Access the WEB Configuration MenuEnd User Password Web PortIP Address PPPoE passwordPPPoE Service Name Dhcp hostnameAllow Dhcp server to Reset TypeMTZ+6MDT+5 Set Time Zone LanguageDND Forward Busy DelayedNAT FXSLayer 3 QoS Admin PasswordFirmware Upgrade Layer 2 QoSHTTP/HTTPS ACS URL Firmware VersionInternet Telephone Service Provider Primary Radius Syslog LevelSend SIP Log Primary Radius AuthRadius Retry Download DeviceDNS Mode Authenticate IDAuthenticate Password Unregister on RebootRegister Expiration Enable Ring-TransferDisable Bellcore Style Registration Retry WaitEnable Call Features SIP T1 TimeoutDisable Dtmf Delay Proxy-RequireDisable Receiver Disable Call WaitingDisable Call-Waiting Disable Reminder RingUse # as Dial Key Dial Plan PrefixDial Plan Dial Plan Rules Callee Request Timer Special FeatureCaller Request Timer Force TimerVAD Disable Line Echo Slic SettingEnable Hook Flash Canceller LECProfile ID Enable PortsSIP Use ID Hunting GroupConfiguration Through a Central Server Saving the Configuration ChangesRebooting the HT70X from Remote Port Sip port that will be annexed to the IP address aboveFirmware Version Software Upgrade Firmware Upgrade Through TFTP/HTTP/HTTPSInstructions for Local Firmware Upgrade Using Tftp Server Instructions for Upload from Local DirectoryFirmware and Configuration File Prefix and Postfix Configuration File DownloadManaging Firmware and Configuration File Download Firmware Version Restore Factory Default Setting Factory ResetReset from web interface Reset Type
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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.