TABLE 10: HT–496 Individual Account Settings Definitions

 

 

 

 

 

 

 

 

 

 

 

SIP Server

 

 

 

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

 

 

 

 

 

 

 

 

 

 

 

Outbound Proxy

 

 

 

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.

 

 

 

 

 

 

 

 

 

SIP User ID

 

 

 

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.

 

 

 

 

 

 

 

 

 

 

 

Authenticate ID

 

 

 

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

 

 

 

 

 

 

 

User ID.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Authenticate Password

 

 

 

SIP service subscriber’s account password.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Name

 

 

 

SIP service subscriber’s name for Caller ID display.

 

 

 

 

 

 

 

 

 

 

 

Use DNS SRV

 

 

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

 

 

 

 

 

 

 

 

User ID is Phone

 

 

 

If the HT–496 has an assigned PSTN telephone number, this field should be

 

 

 

Number

 

 

 

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.

 

 

 

 

 

 

 

 

 

 

 

SIP Registration

 

 

 

Controls whether the HT–496 needs to send REGISTER messages to the

 

 

 

 

 

 

proxy server. The default setting is Yes.

 

 

 

 

 

 

 

 

 

 

 

Unregister on Reboot

 

 

 

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

 

 

 

 

 

 

 

cleared on reboot.

 

 

 

 

 

 

 

 

 

 

 

Register Expiration

 

 

 

This parameter allows the user to specify the time frequency (in minutes) the

 

 

 

 

 

 

 

HT–496 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).

 

 

 

 

 

 

 

 

 

 

 

Local SIP port

 

 

 

Defines the local SIP port the HT–496 will listen and transmit. The default

 

 

 

 

 

 

 

value for FXS port 1 is 5060. The default value for FXS port 2 is 5062.

 

 

 

 

 

 

 

 

 

Local RTP port

 

 

 

Defines the local RTP-RTCP port pair the HT–496 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.

 

 

 

 

 

 

 

 

 

 

 

Use Random Port

 

 

 

This parameter forces the random generation of both the local SIP and RTP

 

 

 

 

 

 

 

ports when set to Yes. This is usually necessary when multiple HT–496 are

 

 

 

 

 

 

 

behind the same NAT.

 

 

 

 

 

 

 

 

 

 

 

DTMF Payload Type

 

 

 

This parameter sets the payload type for DTMF using RFC2833.

 

 

 

 

 

 

 

 

 

Send DTMF

 

 

This parameter specifies the mechanism to transmit DTMF digit. There are 3

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

 

HT–496 User Manual

Page 23 of 33

 

 

 

 

 

 

Firmware 1.0.3.64

Last Updated: 1/2007

 

VoIPon www.voipon.co.uk

sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 600030

Page 23
Image 23
Grandstream Networks user manual HT-496 Individual Account Settings Definitions

HT-496 specifications

Grandstream Networks HT-496 is a cutting-edge Analog Telephone Adapter (ATA) designed to bridge the gap between traditional telephone systems and modern Voice over IP (VoIP) networks. As telecommunications technologies continue to evolve, devices like the HT-496 play a crucial role in ensuring seamless communication across diverse platforms.

One of the main features of the HT-496 is its ability to support up to four simultaneous calls. This is particularly advantageous for small to medium-sized businesses that rely on efficient communication to manage customer interactions and internal coordination. The device comes equipped with two FXS ports, enabling users to connect their existing analog telephones directly, ensuring that they can continue using familiar equipment while benefiting from the advanced features of VoIP technology.

The HT-496 supports a variety of voice codecs, including G.711, G.726, G.729, and G.722, allowing for high-quality audio transmission even in bandwidth-constrained environments. This flexibility ensures users can choose the codec that best fits their specific network conditions, optimizing both call clarity and resource efficiency.

In terms of management and security, Grandstream has integrated several advanced technologies into the HT-496. The device includes support for SIP (Session Initiation Protocol), making it compatible with a wide range of VoIP services. Additionally, it features various security mechanisms, such as SRTP (Secure Real-Time Transport Protocol) and TLS (Transport Layer Security), ensuring that voice communications are encrypted and protected from potential threats.

Installation and configuration of the HT-496 are user-friendly, thanks to its web-based interface. This makes it easy for both technical and non-technical users to manage settings, adjust parameters, and monitor system performance. Furthermore, the device supports automatic provisioning, allowing for quick setup with minimal manual intervention.

Another notable characteristic of the HT-496 is its compact design, which enables easy placement in any office environment. Its durable construction ensures reliable operation over time, making it a cost-effective solution for businesses looking to transition to VoIP technology without discarding their existing analog devices.

In summary, the Grandstream Networks HT-496 features a robust design, compatibility with a variety of voice codecs, advanced security protocols, and user-friendly management options. These characteristics make it an essential tool for businesses seeking to enhance their communication systems while maintaining a connection to traditional telephony. By investing in the HT-496, organizations can simplify their transition to VoIP and unlock the full potential of modern telecommunications.