TABLE 10: HT496 Individual Account Settings Definitions

SIP Server

Outbound Proxy

SIP User ID

Authenticate ID

Authenticate Password

Name

Use DNS SRV

User ID is Phone Number

SIP Registration

Unregister on Reboot

Register Expiration

Local SIP port

Local RTP port

Use Random Port

SIP Registration Failure Retry Wait Time

DTMF Payload Type

Send DTMF

Send Flash Event

This field contains the URI string or the IP address (and port, if different from 5060) of the SIP proxy server. e.g., the following are some valid examples: sip.my-voip- provider.com, or sip:my-company-sip-server.com, or 192.168.1.200:5066

This field contains the URI string or the IP address (and port, if different from 5060) of the outbound proxy. If there is no outbound proxy, this field SHOULD be blank. If not blank, all outgoing requests are sent to this outbound proxy.

This field contains the user part of the SIP address for this phone. e.g., if the SIP address is sip:my_user_id@my_provider.com, then the SIP User ID is: my_user_id.

Do NOT include the preceding “sip:” scheme or the host portion of the SIP address in this field.

SIP service subscriber’s Authenticate ID. Auth ID can be identical to the SIP User ID.

SIP service subscriber’s account password.

SIP service subscriber’s name for Caller ID display.

Default is No. If set to Yes the client will use DNS SRV for server lookup.

If the HT496 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.

Controls whether the HT496 needs to send REGISTER messages to the proxy server. The default setting is Yes.

Default is No. f set to Yes, the SIP user’s registration information will be cleared on reboot.

This parameter allows the user to specify the time frequency (in minutes) the HT496 refreshes its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days).

Defines the local SIP port the HT496 will listen and transmit. The default value for FXS port 1 is 5060. The default value for FXS port 2 is 5062.

Defines the local RTP-RTCP port pair the HT496 will listen and transmit. It is the base RTP port for channel 0. When configured,

channel 0 uses this port _value for RTP and the port_value+1 for its RTCP; channel 1 uses port_value+2 for RTP and port_value+3 for its RTCP.

The default value for FXS port 1 is 5004. The default value for FXS port 2 is 5008.

This parameter forces the random generation of both the local SIP and RTP ports when set to Yes. This is usually necessary when multiple HT496 are behind the same NAT.

In case sip server fail from some reason, the HT496 will try to re-register not according to standard SIP timers, but according to preconfigured interval. Recommended to leave default value 20sec. Designed by special request of big service providers.

This parameter sets the payload type for DTMF using RFC2833.

This parameter specifies the mechanism to transmit DTMF digit. There are 3 modes supported: 1) in audio which means DTMF is combined in audio signal (not very reliable with low-bit-rate codec), 2) via RTP (RFC2833), or 3) via SIP INFO. Multiple selections of DTMF method are supported.

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

Grandstream Networks, Inc.

HT496 User Manual

Page 23 of 29

 

Firmware 1.0.3.86

Last Updated: 7/2007

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Grandstream Networks user manual HT496 Individual Account Settings Definitions

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.