Grandstream Networks HT704, HT702, HT701 user manual Account Settings

Page 24

 

RADIUS Timeout

 

Default is 2. HT704 Only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

RADIUS Retry

 

Default is 3. HT704 Only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Download Device

 

 

 

 

 

Allows user to download and save a text file containing all the P values of each setting as

 

 

Configuration

 

configured at that point on the unit. (Note: For Security Reasons, all Passwords won’t be

 

 

 

 

Downloaded)

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Upload Firmware

 

Allows the user to upgrade the firmware with a single firmware file by browsing and

 

 

 

 

 

 

loading the file from your computer (local directory).

 

 

 

 

 

 

 

 

 

 

Table 11: Account Settings

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Profile Active

 

 

 

When set to Yes the FXS port or Profile (for HT704) is activated.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Primary SIP Server

 

 

 

SIP Server’s IP address or Domain name provided by VoIP service provider.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Failover SIP Server

 

 

 

Failover SIP Server’s IP address or Domain name in case primary server does not

 

 

 

 

 

 

 

 

 

 

respond.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Prefer Primary SIP

 

 

 

Default is No. If set to yes, it will register to Primary Server if registration with Failover

 

 

 

 

 

 

Server

 

 

 

server expires

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Outbound Proxy

 

 

 

IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border

 

 

 

 

 

 

 

 

 

 

Controller. Used by HT70X for firewall or NAT penetration in different network

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

environments. If symmetric NAT is detected, STUN will not work and ONLY outbound

 

 

 

 

 

 

 

 

 

 

 

proxy can correct the problem.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP transport

 

 

 

User can select UDP or TCP or TLS. Default is UDP.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

NAT Traversal (STUN)

 

 

 

This parameter defines whether or not the HT70X NAT traversal mechanism is

 

 

 

 

 

 

 

 

 

 

activated. If activated (by choosing “Yes”) and a STUN server is also specified, then the

 

 

 

 

 

 

 

 

 

 

HT70X performs according to the STUN client specification. Using this mode, the

 

 

 

 

 

 

 

 

 

 

 

embedded STUN client will detect if and what type of firewall/NAT. If the detected NAT

 

 

 

 

 

 

 

 

 

 

 

is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the HT70X will use its

 

 

 

 

 

 

 

 

 

 

 

mapped public IP address and port in all of its SIP and SDP messages.

 

 

 

 

 

 

 

 

 

 

 

If the NAT Traversal field is set to “Yes” with no specified STUN server, the HT70X will

 

 

 

 

 

 

 

 

 

 

 

periodically (every 20 seconds or so) send a blank UDP packet (with no payload data)

 

 

 

 

 

 

 

 

 

 

 

to the SIP server to keep the “hole” on the NAT open.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP User ID

 

 

 

User account information, provided by VoIP service provider (ITSP). Usually in the form

 

 

 

 

 

 

 

 

 

 

 

of digit similar to phone number or actually a phone number. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Authenticate ID

 

 

 

SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or

 

 

 

 

 

 

 

 

 

 

different from SIP User ID. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Authenticate Password

 

 

 

SIP service subscriber’s account password. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Name

 

 

 

SIP service subscriber’s name for Caller ID display. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

DNS Mode

 

 

 

One from the 3 modes are available for “DNS Mode” configuration:

 

 

 

 

 

 

 

 

 

 

 

-A Record (for resolving IP Address of target according to domain name)

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

-SRV (DNS SRV resource records indicates how to find services for various protocols)

 

 

 

 

 

 

 

 

 

 

 

-NAPTR/SRV (Naming Authority Pointer according to RFC 2915)

 

 

 

 

 

 

 

 

 

 

 

One mode can be chosen for the client to look up server.

 

 

 

 

 

 

 

 

 

 

 

 

The default value is “A Record”

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Tel URI

 

 

 

The default setting is “Disabled”. If the phone has an assigned PSTN

 

 

 

 

 

 

 

 

 

 

Number, this field should be set to “User=Phone” then a

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

“User=Phone” parameter will be attached to the “From header” in the SIP

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

 

HT-70X User Manual

Page 24 of 35

 

 

 

 

 

 

 

Firmware Version 1.0.0.18

Last Updated: 03/2012

Image 24
Contents Grandstream Networks, Inc HT701 HT702Onfiguring the HT70X Through V Oice P Rompts Configuration GUI Interface Examples Safety Compliances WarrantyWelcome Connecting the HT70X Connect Your HT70XEquipment Packaging Definitions of the HT70X Connectors Power LEDBasic Definitions of the HT70X LEDs Pattern HT702 HT704Advanced Definitions of the HT70X LEDs Pattern Analog Software Features Overview HT70X Software FeaturesProduct Overview Hardware Specification HT70X Hardware and Technical SpecificationsEMC Basic Operations Understanding HT70X Voice PromptHT70X IVR Menu Definitions Using IVR Using Star CodePlacing a Phone Call ExamplesCall Hold Call WaitingCall Transfer Or Voice Prompt with option 47, then 192*168*0*160Way Conferencing Fax SupportAttended Transfer Instructions for 3-way conferenceCall Features HT70X Call FeaturesEnable Srtp Disable Srtp Blind TransferConfiguration Guide Configuring the HT70X through Voice PromptsConfiguring the HT70X Via Web Browser Important Settings NAT SettingsBasic Settings Status MTZ+6MDT+5NAT Advanced User Configuration Advanced SettingsDND FXSHTTP/HTTPS ACS URL URISyslog Level Primary RadiusAccount Settings Unregister on Reboot Enable Ring-TransferDisable Bellcore Style SIP T1 TimeoutDisable Dtmf Enable Call FeaturesDisable Call Waiting Disable Call-WaitingNo Key Entry Timeout Early DialDial Plan Prefix Use # as Dial KeyVAD Fax Mode Detection ModeSrtp Mode Slic SettingHT704 FXS Ports Settings Saving the Configuration ChangesRebooting the HT70X from Remote Configuration through a Central Server Software Upgrade Firmware Upgrade through TFTP/HTTP/HTTPSInstructions for Upload from Local Directory Instructions for local firmware upgrade using Tftp server Configuration File DownloadFirmware and Configuration File Prefix and Postfix Managing Firmware and Configuration File DownloadYes, every Restore Factory Default Setting Encode the MAC Address
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HT704, HT701, 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.