Grandstream ht-503 Manuel D’Utilisation

 

Grandstream Networks, Inc. 

Page 13 of 37  

1. 

 presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial 

tone. 

2. 

 dials C’s number then # (or wait for 4 seconds).  

3.  If C answers the call, then 

 presses FLASH to bring 

, C in the conference. 

4.  If C does not answer the call, 

 can press FLASH back to talk to 

 

5. If 

 presses FLASH during the conference, C will be dropped out.  

 

Note: Party A is the call initiator for both calls with party B and party C. 

 

HT503 supports PSTN pass through using the FXS port. The user can place and receive PSTN calls 
using analog phone connected to FXS port.   
 

•  To receive PSTN calls, pick up the phone when it rings;    
•  To complete a PSTN call, press the PSTN access code (*00 is default, or any number configured 

in the web configuration) to switch to the PSTN line, listen for a dial tone, then dial the number.  

             It the HT503 loses power, it will function as a jack, enabling a direct connection to the PSTN Line. 

•  If the 503 loses power or lost registration with SIP server, device will switch to mode when PSTN 

line will be transparently connected directly to phone connected to FXS port. It will function as a 
jack, enabling a direct connection to the PSTN Line. 

 

This function is available using the FXO port.  The FXO port functions as a bridge between the Internet 
and PSTN. The user can remotely use a PSTN line to initiate a call.  
 
T

: 

1.  Dial the FXO SIP account phone number to establish the VoIP session. The caller will hear the 

ring back tone once. Then the caller hears either a special continuous tone or a dial tone. The 
special continuous tone is played if the pin code is configured, otherwise, the caller will hear a dial 
tone.  

2.  Enter the pin code (configured on the configuration page). The caller will hear a dial tone and be 

connected to the PSTN line if the pin code is valid.  If the pin code is invalid, the continuous tone 
is played to prompt caller to enter the pin code again. The user may try up to 3 times to enter a 
correct pin code.  After three (3) tries, the HT503 hangs up. 

3.  After the caller hears a dial tone from PSTN line, the caller can place the next call.   

4.  The user can hit the # key to identify the end of the pin code or wait 4 seconds for a new dial tone 

and then

 

dialing the PSTN number. 

 
Note:   

•  Users can choose whether or not to apply password protection for VoIP-to-PSTN calls. A PIN 

(Pin for PSTN calls) consists of up to 8 numeric digits and can be configured using the BASIC 
SETTINGS of the web configuration page.  By default, there is no password protection. (I.e. there 
is no authentication required for callers on the use of PSTN line through HT503).   

•  When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the HT503 

FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission.  

•  The special continuous tone is the prompt to enter a valid PIN code.  If a caller doesn’t enter a 

valid PIN, the HT503 times out after 10 seconds. Users may press the “#” key to indicate the end 
of an input or wait 4 seconds.