Ht503 User Manual - Grandstream Networks HT-503 User Manual

Fxs/fxo port analog telephone adaptor

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User manual (85 pages)

Setup (11 pages)

Installing and configuring (3 pages)

Table Of Contents

Table of Contents

SIP T2 Interval

DTMF Payload Type

DTMF in Audio

DTMF via RFC2833

DTMF via SIP INFO

Send Flash Event

Enable Call Features

Proxy Require

Use NAT IP

Disable Call Waiting

Tone

Ring Timeout

Early Dial

Dial Plan Prefix

Use # as Dial Key

Dian Plan

Subscribe for MWI

Grandstream Networks, Inc.

Maximum retransmission interval for non-INVITE requests and INVITE responses.

Sends DTMF using RFC2833

Sends DTMF as inband (in-audio).

Sends DTMF via RTP (according the RFC2833).

Send DTMF as a SIP INFO message.

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

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

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

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

Turns off the Call Waiting feature. Default is No.

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

CWCID will still be displayed.

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

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.

Sets the prefix added to each dialed number.

This allows users to configure the # key as the "Send" (or "Dial") key. If set to "Yes", "#"

will send the number. In this case, this key is essentially equivalent to the "Dial" key. If

set to "No", the "#" key can be included as part of a number.

Dial plans work only for incoming calls from PSTN network. In case unconditional call

forward to VoIP is configured, dial plan feature will not work. In case of normal dialing

to VoIP, after dialing PSTN number,

If using the 'hop on/hop off' feature, the dial plan rules affect only the last called number

(i.e. the number called after receiving dial tone from the ATA).

Dial Plan Rules:

Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d

Grammar: x - any digit from 0-9;

a. xx+ - at least 2 digits number;

b. xx. – at least 2 digits number;

c. ^ - exclude;

d. [3-5] - any digit of 3, 4, or 5;

e. [147] - any digit 1, 4, or 7;

f. <2=011> - replace digit 2 with 011 when dialing

•

Example 1: {[369]11 | 1617xxxxxxx} –

Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617

Example 2: {^1900x+ | <=1617>xxxxxxx} –

•

Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit

numbers

Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –

•

Allow any length of number with leading digit 2 and 10 digit-numbers of leading

digit 1 and leading exchange number between 2 and 9; If leading digit is 2,

replace leading digit 2 with 011 before dialing.

Default: Outgoing - {x+}

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