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Grandstream Networks, Inc., 1297 Beacon St., 2
nd
Flr, Brookline, MA 02446 USA
Tel: (617) 566-9300, FAX: (617) 249-1987
www.grandstream.com
METHOD 2
Configure GXW410x to function as a PEER
gateway
(No SIP accounts needed; only used for one-stage
dialing)
METHOD 1 (Configuration):
Please follow the steps below:
1. Enter a SIP Server IP Address (or FQDN ex. sip.mysipserver.com) in the SIP Server field under
Profile 1 page.
2. Go to the Channels page and enter up to 8 accounts/extensions along with their Authentication ID
and Authentication Passwords, and select the corresponding Profile.
Channels SIP
User
ID
Authen
ID
Authen
Password
Profile ID
1 101 101 *** Profile1
2 102 102 *** Profile1
3 103 103 *** Profile1
4 104 104 *** Profile1
5 105 105 *** Profile1
6 106 106 *** Profile1
7 107 107 *** Profile1
8 108 108 *** Profile1
3. One of the most important settings on the GXW410x is the “Stage Method” setting under FXO
Lines page. You can set this different (1 or 2) for each Channel. For simplicity purposes please keep
Stage Method to 2.
Ex. Stage Method (1/2): Ch1-8:2;
4. Click on Update and Reboot the unit. When the GXW410x boots up, click on “Status Page” to
check if the Account/extensions show up as “Registered:Yes”. If not, please double check the passwords
or network settings.
5. Once the accounts are registered to the SIP Server, you should be able to make VoIP to PSTN
calls (assuming you have physical PSTN Lines connected to the FXO Ports at the back panel of the unit).
Below is an example showing VoIP-to-PSTN call flow:
y Accounts 101 to 108 registered on the Channels page to SIP Server A.
y IP Phone with Account 201 registered to the same SIP Server A.
y PSTN Line X is connected to FXO X port on GXW-410x.
a. 201 dials 101 (or 102/103/104/105/106/107/108).
b. It will hear ring back tone and then PSTN dial tone played from PSTN Line X.
c. 201 can dial out via PSTN Line X.
6. PSTN-to-VoIP call flow:
a. PSTN number Y dials PSTN number X (connected to FXO1 on the gateway)
b. Y gets ring back tone and then VoIP dial tone played from 101 (only)
Note: VoIP-to-PSTN calls function in round robin fashion, so the next available port is selected to route
the call. PSTN-to-VoIP calls depend on the PSTN line you are calling and will be routed to the
corresponding VoIP Account on that channel.
Last updated on April 1, 2008
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