HandyTone-386 User Manual

Grandstream Networks, Inc.

User ID is Phone Number

Early Dial

Dial Plan Prefix

No Key Entry

Timeout

Use # as

Send Key

Use Random Port

NAT Traversal

Keep-alive interval

Use NAT IP:

Proxy-Require

TFTP Upgrade Server

If the HandyTone ATA has an assigned PSTN telephone number, this field should be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP request

Default is No. Use only if proxy supports 484 response

Sets the prefix added to each dialed number

Default is 4 seconds.

This parameter allows users to configure the “#” key to be used as the “Send” (or “Dial”) key. If set to “Yes”, pressing this key will immediately trigger the sending of dialed string collected so far. In this case, this key is essentially equivalent to the “(Re)Dial” key. If set to “No”, this “#” key will then be included as part of the dial string to be sent out.

This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple HandyTone ATAs are behind the same NAT.

This parameter defines whether the HandyTone ATA NAT traversal mechanism will be activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, then the HandyTone ATA will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the HandyTone ATA will attempt to detect if and what type of firewall/NAT it is sitting behind through communication with the specified STUN server. If the detected NAT is a Full Cone, Restricted Cone, or a Port- Restricted Cone, the HandyTone ATA will attempt to use its mapped public IP address and port in all its SIP and SDP messages. If the NAT Traversal field is set to “Yes” with no specified STUN server, the HandyTone ATA 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.

This parameter specifies how often the HandyTone ATA sends a blank UDP packet to the SIP server in order to keep the “hole” on the NAT open.

NAT IP address used in SIP/SDP message. Default is blank.

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

This is the IP address of the configured TFTP server. If it is non-zero or not blank, the HandyTone ATA will attempt to retrieve new configuration file or new code image from the specified TFTP server at boot time. It will make up to 3 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code image is retrieved, the new downloaded image will be verified and then saved into the Flash memory.

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Grandstream Networks Handy Tone 386 user manual Grandstream Networks, Inc

Handy Tone 386 specifications

The Grandstream Networks Handy Tone 386 is a robust VoIP adapter designed for both residential and small business environments, providing an effective way to connect traditional telephones to Voice over IP networks. This device offers a reliable gateway that allows users to leverage the advantages of modern communication technology while using familiar equipment.

One key feature of the Handy Tone 386 is its support for two FXS ports. These ports enable users to connect standard analog phones directly to the adapter, effectively converting analog voice signals into digital data that can be transmitted over the internet. This allows for seamless communication without the need to invest in new hardware, making it a cost-effective solution for many users.

Another significant aspect is the device's ability to support both SIP and multicast SIP protocols, ensuring compatibility with a wide range of VoIP service providers. By offering multiple protocols, the Handy Tone 386 can provide flexibility in terms of service choices. Users are not locked into a single provider and can easily switch services if needed.

The Handy Tone 386 also comes equipped with advanced technologies such as QoS (Quality of Service) features. This functionality prioritizes voice traffic over other types of data, which is crucial for maintaining call clarity and reducing latency during voice communications. The implementation of secure encryption protocols, such as TLS and SRTP, further ensures that calls are secure and safe from potential eavesdropping or tampering.

With built-in NAT traversal capabilities, the Handy Tone 386 can handle complex network configurations, enabling easy integration into various home or business broadband setups. This makes for straightforward installation and usability, ensuring that users can quickly get up and running without extensive technical knowledge.

The device also includes an intuitive web-based user interface, allowing users to manage settings and configurations easily. This interface facilitates remote management, enabling adjustments to be made without requiring physical access to the unit.

In summary, the Grandstream Networks Handy Tone 386 is a versatile and powerful VoIP adapter. Its dual FXS ports, support for multiple VoIP protocols, QoS features, and security advancements make it a strong choice for users looking to transition to VoIP communication while maximizing their existing telephone infrastructure.