Grandstream Networks GXW-410x Quick Installation Manual page 5

3. One of the most important settings on the GXW410x is the "Stage Method" setting under
FXO Lines webpage. You can set this different (1 or 2) for each Channel. For simplicity,
keep Stage Method to 2. Ex. Stage Method (1/2) : Ch1-8:2
4. Now, click on Update and reboot the unit. When it boots up, click on Status webpage to
check if the Account/extensions show up as "Registered:Yes". If not, ensure the passwords
or network settings are correct.
5. Next, place a VOIP to PSTN call (accounts must be registered to your SIP server and your
physical PSTN Lines are connected to the FXO Ports at the back panel of the unit).
Example of a VOIP to PSTN call flow:
•
Accounts 101 to 108 registered on the Channels webpage to the SIP Server A.
•
IP Phone with Account 201 is registered to same SIP Server A
•
PSTN Line X is connected to FXO1 port on the gateway unit.
a. 201 dials 101 (or 102/103/104/105/106/107/108)
b. You will hear ringback and then PSTN dialtone played from PSTN Line X
c. Now, 201 can dial out onto PSTN Line X
Example of a PSTN to VOIP call flow:
•
PSTN number Y dials PSTN number X (connected to FXO1 on the gateway)
•
You will hear ring-back and then VOIP dial-tone played from 101 (only)
Note One: VOIP to PSTN calls function in round robin method, 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.
Note Two: For two-stage dialing, the "Wait for Dial Tone" field under FXO Lines webpage
should be set to N (No) Ex. Wait for Dial Tone: ch1-8:N. For one stage dialing, configure
the SIP Server to forward certain calls (ex. external calls dialed with a prefix, etc) to the
gateway, before you configure the gateway.
6. On the gateway, change the Stage Method field to 1. ex. Ch1-8:1
7. Now, based on the prefix you have configured on the SIP Server (ex. 91), if a user dials
any number 91-xxx-xxxx, it will be routed to the gateway, which will route it to the PSTN
directly.
8. Next you need to enter a VOIP account number for Off-hook Auto Dial (VOIP) under FXO
Lines webpage. This account generally will be a dummy account on your SIP Server which
is used as a forwarding number for all incoming PSTN calls on the gateway. Ex. Off-hook
Auto Dial (VOIP) : ch1-8:500. Now, all incoming PSTN calls will directly be forwarded to
500 on SIP Server A.
Note: In regions other than North America, the user is also required to configure call progress
tones and PSTN line termination fields. Check with local PSTN service carriers on values
service providers use on the lines. If service provider does not provide those values and users
don't know the correct values, please use the default values. Please contact product support if
you still have questions about configuring your GXW410x.
www.grandstream.com info@grandstream.com GXW-410x Quick Install Guide, p. 5