Grandstream Networks HT496 user manual Last Updated 7/2007

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Enable Call Features

Use Bell-style

3-way Conference

Off-hook Auto-Dial

Proxy-Require

Disable Call Waiting

Disable Call Waiting Caller ID (CWCID)

NAT Traversal (STUN)

No Key Entry Timeout

Preferred Vocoder

Voice Frames per TX

G723 Rate:

iLBC frame size:

iLBC payload type:

Silence Suppression

Fax Mode

Default is Yes. Advanced call features and feature codes functions are supported locally.

If this parameter is set to “Yes”, user will be able to make Bellcore style 3-way conference. *23 will be disabled.

This parameter automatically configures and dials User ID or extension number upon off-hook. Only the user part of a SIP address needs is entered here. The HT496 will automatically append the “@” and the host portion of the corresponding SIP address.

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

Default is No. User can use * code to use this feature per call basis.

Default is No. If set to YES, user will not be able to see CWCID information. If set to No, also requires support of this feature by analog phone connected to FXS port.

This parameter defines whether the HT496 NAT traversal mechanism is activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, then the HT496 performs according to the STUN client specification. Under this mode, the embedded STUN client will detect if and what type of firewall/NAT is used. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the HT496 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 HT496 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.

Default is 4 seconds. While dealing, the number will considered as completed if during this timeout no additional entry has been done.

The HT496 supports up to 6 different Vocoder types including G.711 A-/U-lawG.723.1, G.726, G.729A/B, iLBC.

Users can configure Vocoders in a preference list included with the same preference order in SDP message. The first Vocoder in this list is the appropriate option in “Choice 1”. The last Vocoder in this list is the appropriate option in “Choice 6”.

Number of voice frames transmitted in a single packet. User should be aware of the requested packet time (used in SDP message). This parameter is associated with the first vocoder in the Vocoder Preference List or the actual payload type negotiated between the 2 conversation parties at run time.

e.g. if the first vocoder is configured as G.723 and the “Voice Frames per TX” is set to 2, then the “ptime” value in the SDP message of an INVITE request will be 60ms because each G.723 voice frame contains 30ms of audio.

If the configured voice frames per TX exceeds the maximum allowed value, the HT496 will use and save the maximum allowed value for the corresponding first vocoder choice. The maximum value for PCM is 10(x10ms) frames:

for G.726, it is 20 (x10ms) frames; .

for G.723, it is 32 (x30ms) frames;

for G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames respectively.

Default is 6.3kbps rate. Defines the encoding rate for G723 vocoder.

Sets the iLBC frame size in 20ms or 30ms

Defines payload time for iLBC. Default value is 97. Valid range: 96 - 127.

This controls the silence suppression/VAD feature of G.723. If set to “Yes”, a small quantity of VAD packets (instead of audio packets) will be sent during the a period of silence. If set to “No”, feature is disabled.

T.38 (Auto Detect) FoIP (fax over IP) by default, or Pass-Through (must use codec PCMU/PCMA)

Grandstream Networks, Inc.

HT496 User Manual

Page 24 of 29

 

Firmware 1.0.3.86

Last Updated: 7/2007

Image 24
Contents Grandstream Networks, Inc Onfiguration Through a C Entral S Erver Configuration GUI Interface Examples Igure 3 S Creenshot of C Onfiguration L OG in P AGEWelcome Installation Connecting Your ATADefinitions of the HT496 Connectors Equipment PackagingFive Easy Steps to Install the HT496 Configuring the HT496KEY Features HT496 Technical SpecificationsProduct Overview HT496 Hardware Specifications LEDBasic Operations GET Familiar with Voice PromptHT496 IVR Menu Definitions Main MenuIVR Definition Notes Placing a Phone Call Call HoldCall Waiting Call TransferExpected outcomes WAY ConferencingCall Features HT496 Call Feature DefinitionsLED Light Pattern Indication Able 6 HT496 LED DefinitionsConfiguration Guide Configuring HT496 Through Voice Prompt Dhcp ModeStatic IP Mode Configuring HT496 with WEB Browser END User Configuration User Level Password Web pages allowedHT496 Basic Configuration Settings Definitions Device Mode AccessDaylight Savings Time WAN side HTTP/TelnetAdvanced User Configuration HT496 Device Status Page DefinitionsHT496 Advanced Configuration Page Definitions Caller ID Scheme On-hook VoltagePolarity Reversal NTP serverHT496 Individual Account Settings Definitions Last Updated 7/2007 Audible Tones Saving the Configuration Changes Rebooting the HT496 from RemoteConfiguration Through a Central Server Software Upgrade Firmware Upgrade Through TFTP/HTTPConfiguration File Download Firmware and Configuration File Prefix and PostfixManaging Firmware and Configuration File Download Restore Factory Default Setting Reset VIA IVR
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HT496 specifications

Grandstream Networks has established itself as a leading provider of communication solutions, specializing in voice-over-IP (VoIP) technology. Among its diverse product line, the HT386, HT496, HT488, HT486, and HT286 analog telephone adapters stand out as exceptional devices tailored for seamless integration into modern telecommunication systems.

The Grandstream HT386 is particularly noted for its robust performance and versatility. It supports up to 4 lines, making it ideal for small to medium-sized businesses that require efficiency and reliability. The unit features advanced security protocols such as SRTP and TLS to protect voice communications, ensuring that data is secure during transmission. Additionally, the HT386 boasts an easy installation process and web-based management, which simplifies configuration and maintenance.

Next in line is the HT496, which caters to users with even more demanding requirements. This adapter accommodates up to 4 FXS ports, allowing the connection of multiple analog devices. Enhanced features like 2 SIP accounts and high-definition voice codecs ensure clear audio quality. The HT496 is designed to offer seamless interoperability with various IP routers and switches, making it a flexible solution for businesses expanding their communication framework.

The HT488, another notable entry, is geared towards those looking for high-performance analog telephony. With support for 2 lines and advanced echo cancellation technologies, it guarantees crystal-clear calls, minimizing disruptions during conversations. Additionally, it provides multiple network connectivity options, including DHCP and static IP, allowing users to choose the best configuration suitable for their network environment.

The HT486 offers similar benefits but is optimized for users who require a compact solution. This model features an elegant design while maintaining support for essential VoIP features. With 2 FXS ports and built-in firewall capabilities, it ensures secure and efficient communication for residential and small business users.

Finally, the HT286 is perfect for those seeking an entry-level adapter without compromising on quality. Supporting a single line with a straightforward setup process, it is ideal for users transitioning from traditional phone systems to VoIP technology. This model is also compatible with various VoIP service providers, ensuring users have flexibility when choosing their phone services.

In summary, Grandstream’s HT series—HT386, HT496, HT488, HT486, and HT286—delivers a comprehensive range of features and technologies suited for different communication needs. Each model combines quality with user-friendly interfaces, ensuring that users can fully leverage the benefits of VoIP, whether for personal or business use.