Grandstream Networks HT503 FXS/FXO

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 to look up the server.

User ID is Phone Number If the HT503 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.

SIP Registration Controls whether the HT503 needs to send REGISTER messages to the proxy server.

The default setting is “Yes.”

Unregister on Reboot Default is No. If set to Yes, the SIP user’s registration information will be cleared on

reboot.

Outgoing Call Without

Registration Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if

allowed by ITSP) but is unable to receive incoming calls.

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 HT503 will listen and transmit. The default value for FXS

port is 5062.

Local RTP Port Defines the local RTP-RTCP port pair the HT503 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 5012.

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 HT503 units are behind the same

NAT.

Refer to Use Target

Contact Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the

transferred target’s contact header information.

DTMF Payload Type Sends DTMF using RFC2833

DTMF in Audio Sends DTMF as inband (in-audio).

DTMF via RFC2833 Sends DTMF via RTP (according the RFC2833).

DTMF via SIP INFO Send DTMF as a SIP INFO message.

Send Flash Event Default is No. If set to “Yes”, flash will be sent as DTMF event

Enable Call Features Default is Yes. Toggles support for advanced call features and star code functions.

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

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

Distinctive Ring Tone Customizes Ring Tones 1 to 3 with an associated Caller ID. When selected, the device

will ONLY use this ring tone when the incoming call is set up for Caller ID. The System

Ring Tone is used for all other calls. When selected with no Caller ID configured, the

selected ring tone will be used for all incoming calls.

Disable Call Waiting T urns off the Call Waiting feat ure. Default is No.

Disable Call Waiting

Tone Default is No. This is to disable the stutter Call Waiting Tone during a CWC. The

CWCID will still be displayed.

Ring Timeout Sets the time in which an incoming call will stop ringing when not picked up.

Early Dial Default is No. Use only if proxy supports 484 response. This parameter controls

whether the phone will send an early INVITE each time a key is pressed when a user

dials a number. If set to “Yes”, an INVITE is sent using the dial-number collected thus

far. Otherwise, no INVITE is sent until the “(Re-)Dial” button is pressed or after about 5

seconds have elapsed. The “Yes” option should be used ONLY if there is a SIP proxy

configured and the proxy server supports 484 Incomplete Address response.

Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error).

Note: This feature is NOT designed to work with and should NOT be enabled for

direct IP-to-IP calling.

Grandstream Networks, Inc. HT503 User Manual Page 27 of 35

Firmware 1.0.0.6 Last Updated: 6/2007