Grandstream ht-503 Manuel D’Utilisation

 

Grandstream Networks, Inc. 

Page 14 of 37  

•  On the web configuration page, if the “Forward to PSTN” is configured, the second stage dialing 

format is eliminated, so after dialing into the FXO SIP account number, the PSTN number will be 
called automatically 

This function is available using the FXO port. The FXO port functions as a bridge between the Internet 
and PSTN and enables calls to be passed from the PSTN network to VoIP. The user can make VoIP calls 
remotely by dialing into the FXO line port on HT503.  
 
T

: 

1.  Make an incoming call to the PSTN line on FXO port. The phone will ring for 4 times by default 

(this setting is configurable on the configuration page). 

2.  If no one answers the call after 4 rings (default configuration), then the caller hears either a 

special continuous tone (prompting a PIN number) or a dial tone. 

3.  Enter a valid PIN.  The caller will hear dial tone and be bridged to VoIP.  If an incorrect PIN is 

input, the continuous tone prompts caller to enter a valid PIN.  The caller may try 3 times to enter 
a valid PIN, if it is invalid the HT503 will hang up. 

4.  The caller can dial a VoIP number followed by # (or wait for 4 seconds); the VoIP call will be 

initiated from the SIP account configured on the FXO port.  

 
N

•  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.  

•  On the web configuration page, if the “Forward to VoIP” is configured, the second stage dialing 

format is eliminated, so after dialing into the FXO SIP account number, the PSTN number will be 
called automatically 

 

The FXO port enables access to the PSTN network.  By default, the HT503 is in VoIP mode at off-hook.  
If “Route Call to PSTN” is configured, certain calls will be initiated from the FXO PSTN line port.  This call 
feature is especially useful for emergency calls or local telephone calls.  
 
To use this feature, users need to specify a special rule using the dial plan parameter located under FXS 
Port configuration page. If the dialed digits match the specified prefix, outbound calls will be initiated from 
the PSTN line. 
 
Note: The route to PSTN feature is only applicable to a phone connected to the FXS Port.  The 
configuration is done using the “dial plan” feature under the FXS tab.  An example of the configuration is 
{L: 911x+}. This shows that only calls that start with 911 are immediately forwarded to the PSTN line.  All 
other numbers will not be routed to the PSTN.  An normal # would be: {L: 617x+|x+} or {x+| L: 617x+} 
 
For example, if “Route Call to PSTN” is configured as {L: 626x+}, all outgoing calls starting with 626 will 
be initiated from the PSTN line.