incoming INVITE

 

call will be rejected. If this option is enabled, the device will not be able to make direct

 

 

 

 

 

IP calls.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Allow Incoming SIP

 

Default is No. Check the incoming SIP messages. If they don’t come from the SIP

 

 

 

Messages from SIP

 

proxy, they will be rejected. If this option is enabled, the device will not be able to make

 

 

 

Proxy Only

 

direct IP calls.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP T1 Timeout

 

T1 is an estimate of the round-trip time between the client and server transactions.

 

 

 

 

 

If the network latency is high, select larger value for more reliable usage. Default is 0.5

 

 

 

 

 

Sec.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP T2 Interval

 

Maximum retransmission interval for non-INVITE requests and INVITE responses.

 

 

 

 

 

Default is 4 Sec.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

DTMF Payload Type

 

Sets the payload type for DTMF using RFC2833. Default is 101.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Preferred DTMF method

 

The HT70X supports up to 3 different DTMF methods including in-audio, via RTP

 

 

 

 

 

(RFC2833) and via Sip Info using SIP INFO messages. The user can configure DTMF

 

 

 

 

 

method in a priority list.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Disable DTMF

 

Default is No. If set to yes, use above DTMF order without negotiation

 

 

 

 

Negotiation

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Send Flash Event

 

Default is No. If set to yes, flash will be sent as DTMF event.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Enable Call Features

 

Default is Yes. (If Yes, call features using star codes will be supported locally)

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Offhook Auto-Dial

 

This parameter allows users to configure a User ID or extension number that is

 

 

 

 

 

automatically dialed when off-hook. Only the user part of a SIP address needs to be

 

 

 

 

 

entered here. The HT70X will automatically append the “@” and the host portion of the

 

 

 

 

 

corresponding SIP address. HT701 and HT702 only

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Offhook Auto-Dial

 

The auto-dial delay after off hook.

 

 

 

 

Delay

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Proxy-Require

 

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

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Use NAT IP

 

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

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Use SIP User-Agent

 

Configurable SIP User-Agent Header.

 

 

 

 

Header

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Distinctive Ring Tone

 

Custom Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is

 

 

 

 

 

configured, then the device will ONLY uses this ring tone when the incoming call is

 

 

 

 

 

from the Caller ID. System Ring Tone is used for all other calls. When selected but no

 

 

 

 

 

Caller ID is configured, the selected ring tone will be used for all incoming calls using

 

 

 

 

 

the FXS port or Profile. Distinctive ring tones can be configured not only for matching a

 

 

 

 

 

whole number, but also for matching prefixes. In this case symbol * (star) will be used.

 

 

 

 

 

For example:

 

 

 

 

 

 

 

if configured as *617, Ring Tone 1 will be used in case of call arrived from the area

 

 

 

 

 

code 617. Any other incoming call will ring using cadence defined in parameter System

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

FIRMWARE VERSION 1.0.3.1

HT70X USER MANUAL

Page 38 of 52

 

Page 38
Image 38
Grandstream Networks HT704 Incoming Invite, Allow Incoming SIP, Messages from SIP, Proxy Only, SIP T1 Timeout, Use NAT IP

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.