Grandstream Networks GXW-410xv Application Description

Application Description

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.

FIGURE 3: FUNCTIONAL DIAGRAM OF IP-PBX & GXW-410XV

Anywhere in the world

GXW-410xv

4 or 8 Ports FXO Lines

PSTN Analog

Grandstream IP Phones

IPPBX or

SIP Server

IP/LAN IP/WAN

SIMPLE CONFIGURATION: MEDIA GATEWAY TO ACCESS PSTN NETWORKS

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 Page 7 of 30

Firmware 1.0.0.36 Updated: 11/2006