Grandstream Networks GXW410X METHOD 1 Configuration, Channels, Authen, Profile ID, User, Password

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	Authen	Authen	Profile ID
	User	ID	Password
	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:

yAccounts 101 to 108 registered on the Channels page to SIP Server A.

yIP Phone with Account 201 registered to the same SIP Server A.

yPSTN 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.

Grandstream Networks, Inc., 1297 Beacon St., 2nd Flr, Brookline, MA 02446 USA

Tel: (617) 566-9300, FAX: (617) 249-1987

www.grandstream.com Last updated on April 1, 2008

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