Справочник Пользователя для Grandstream ht-503

 

Grandstream Networks, Inc. 

Page 29 of 37  

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: 

1. 

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

2. 

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. 

3. 

Default: Outgoing - {x+} 

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