Grandstream Networks GXW-410xv Manuel D’Utilisation

A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-410xv 
series. In this environment, the SIP server handles SIP registration and call control and the GXW-410xv 
processes media conversion between IP and PSTN calls. By design, the system supports the call 
progress tones and PSTN signaling standards for North America, Europe, Latin America, Asia, and 
various other countries/regions.   
 
 

 
 

IPPBX or  

SIP Server 

IP/LAN 

IP/WAN 

 

 
 

GXW-410xv can be configured to work with any leading SIP server, for a pure media gateway to access 
PSTN networks. In such applications, the user only needs to configure GXW-410xvgateway Stage Dialing 
field and Sip Server field. 
 
For a simple set-up, users only need to configure a SIP server field for default SIP Profile 1. This field 
should be configured to point to the SIP server to be used with the GXW-410xv.  
 
For advanced applications, the user is required to choose at least one SIP server field from the SIP 
profiles and one stage dialing under system Channel configuration table.  On SIP server sides, the SIP 
server must be configured to forward user PSTN calls to the GXW-410xv.  
 
Please be aware that by default, the system uses North American PSTN settings and TWO STAGE 
dialing to access PSTN networks for VOIP to PSTN calls and PSTN to VOIP calls. Two stage dialing 
means the end-user will hear dial-tone twice. First dial-tone is used to let users to input destination 
number in the same network of the calling networks. Second dial-tone is used to let users to input final 
destination number.  
 

Grandstream Networks, Inc. 

GXW-410xv User’s Manual 

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