Grandstream Networks GXW-410xv - Manual (Page 7)

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.
F IGURE 3: F UNCTIONAL D IAGRAM OF IP-PBX & GXW-410 XV
P P S S T T N N
C C l l o o u u d d
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
S
IMPLE
C
ONFIGURATION
: M
EDIA
G
ATEWAY TO
A
CCESS
PSTN N
ETWORKS
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