Subscribe for MWI

 

 

 

Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication will be

 

 

 

 

 

 

 

 

 

 

 

sent periodically.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Send Anonymous

 

 

 

Default is No. If this parameter is set to “Yes”, the “From” header along with Privacy

 

 

 

 

 

 

 

 

 

and P_ Asserted_Identity headers in outgoing INVITE message will be set to

 

 

 

 

 

 

 

 

 

anonymous, blocking Caller ID.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Anonymous Call

 

 

 

Default is No. If set to Yes, incoming calls with anonymous Caller ID will be rejected

 

 

 

 

 

Rejection

 

 

 

with 486 Busy message.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Special Feature

 

 

 

Default is Standard. Choose the selection to meet some special requirements from

 

 

 

 

 

 

 

 

 

 

 

Softswitch vendors.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Session Expiration

 

 

 

Grandstream implemented SIP Session Timer. The session timer extension enables

 

 

 

 

 

 

 

 

 

SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE.

 

 

 

 

 

 

 

 

 

Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE

 

 

 

 

 

 

 

 

 

 

 

message, the session will be terminated. Session Expiration is the time (in seconds) at

 

 

 

 

 

 

 

 

 

 

 

which the session is considered timed out, if no successful session refresh transaction

 

 

 

 

 

 

 

 

 

 

 

occurs beforehand. The default value is 180 seconds.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Min-SE

 

 

 

The minimum session expiration (in seconds). The default value is 90 seconds.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Caller Request Timer

 

 

 

Default is No. If selecting “Yes” the phone will use session timer when it makes

 

 

 

 

 

 

 

 

 

 

 

outbound calls if remote party supports session timer.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Callee Request Timer

 

 

 

Default is No. If selecting “Yes” the phone will use session timer when it receives

 

 

 

 

 

 

 

 

 

inbound calls with session timer request.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Force Timer

 

 

 

Default is No. If selecting “Yes” the phone will use session timer even if the remote

 

 

 

 

 

 

 

 

 

party does not support this feature. Selecting “No” will allow the phone to enable

 

 

 

 

 

 

 

 

 

session timer only when the remote party support this feature. To turn off Session

 

 

 

 

 

 

 

 

 

 

 

Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

UAC Specify Refresher

 

 

 

Default is Omit. As a Caller, select UAC to use the phone as the refresher, or UAS to

 

 

 

 

 

 

 

 

 

use the Callee or proxy server as the refresher.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

UAS Specify Refresher

 

 

 

Default is UAC. As a Callee, select UAC to use caller or proxy server as the refresher,

 

 

 

 

 

 

 

 

 

or UAS to use the phone as the refresher.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Force INVITE

 

 

 

Session Timer can be refreshed using INVITE method or UPDATE method. Select

 

 

 

 

 

 

 

 

 

“Yes” to use INVITE method to refresh the session timer. Default is No,

 

 

 

 

 

 

 

 

 

 

 

 

Send Re-INVITE After

 

 

 

Default is No, If set to “Yes”, device will send an INVITE with audio vocoders upon

 

 

 

 

 

 

 

Fax

 

 

 

completion of Fax to continue session in audio only.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Enable Silence

 

 

 

For fax machines that do not send a Disconnect when fax is done. This option

 

 

 

 

 

Detection for Fax

 

 

 

Enables/Disables the detection of silence in order to know the fax has finished. The

 

 

 

 

 

Disconnect

 

 

 

silence period is non-configurable and fixed to 7 seconds. Default is No,

 

 

 

 

 

 

 

 

 

 

 

 

 

Enable 100rel

 

 

 

Default is No, If set to Yes, Enables the use of PRACK (Provisional Acknowledgment)

 

 

 

 

 

 

 

 

 

method.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Use First Matching

 

 

 

Default is No. If set to “Yes”, device will include only the first match vocoder in its

 

 

 

 

 

 

 

Vocoder in 200OK SDP

 

 

 

200OK response, otherwise it will include all match vocoders in same order received in

 

 

 

 

 

 

 

 

 

INVITE.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Preferred Vocoder

 

 

 

The HT70X supports up to 5 different Vocoder types including G.711 A-/U-law, G.726-

 

 

 

 

 

 

 

 

 

32, G.723.1, G.729A/B/E, iLBC. The user can configure Vocoders in a preference list

 

 

 

 

 

 

 

 

 

that will be included with the same preference order in SDP message. The first

 

 

 

 

 

 

 

 

 

Vocoder is entered by choosing the appropriate option in “Choice 1”. The last Vocoder

 

 

 

 

 

 

 

 

 

 

 

is entered by choosing the appropriate option in “Choice 8”.

 

 

 

 

 

 

 

 

 

 

 

 

Vocoder types can also be changed per call basis by using a star code. Please see the

 

 

 

 

 

 

 

 

 

 

 

“Call features” section.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

G723 Rate

 

 

 

Default is 6.3kbps. Defines the encoding rate for G.723.1 vocoder.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

iLBC Frame Size

 

 

 

Default is 20ms. Sets the iLBC frame size in 20ms or 30ms.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

iLBC Payload type

 

 

 

Defines payload type for iLBC. Default value is 97. The valid range is between 96 and

 

 

 

 

 

 

 

 

 

127.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

VAD

 

 

 

Default is No. VAD allows detecting the absence of audio and conserve bandwidth by

 

 

 

 

 

 

 

 

 

preventing the transmission of "silent packets" over the network.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

 

HT-70X User Manual

Page 28 of 35

 

 

 

 

 

 

 

Firmware Version 1.0.0.18

Last Updated: 03/2012

Page 28
Image 28
Grandstream Networks HT702, HT704, HT701 user manual Vad

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.