Grandstream Networks, Inc. GXW410x User Manual Page 20 of 35
Firmware Version 1.3.4.13 Last Updated: 3/2012
AC Termination
Impedance
Selects the impedance of the analog line connected to the FXO port on the GXW410x. Here is
some basic information which may be helpful for initial configuration:
600 Ohm – North America;
270 Ohm + (750 Ohm || 150 nF) -- Most of Europe
220 Ohm + (820 Ohm || 120 nF) – Australia, New Zealand
220 Ohm + (820 Ohm || 115 nF) – Austria, Bulgaria, Germany, Slovakia, South Africa
370 Ohm + (620 Ohm || 310 nF) – UK., India
If this parameter is not configured properly you may experience echo or static in the line. Please
refer to Table 9 (FXO Lines Test Tab Definition). This tool will run an automated test to
determine the correct impedance value to match your lines
Wait for Dial-tone
Default is Yes. When set to Yes, the gateway will recognize dial-tone from the Central Office
(CO) before it completes call. If you can’t make an outbound call, set this is No.
Stage Method
Syntax - ch1-8:1; {all channels 1 to 8 are set to value 1 or 2}
Stage method can be set to either 1 or 2.
Set this parameter to 1 if you need to make a direct PSTN call from a VOIP endpoint. When you
set it to 2, you will first dial one of the VOIP channel accounts from the VOIP endpoint, this will
result in getting a PSTN line dial-tone to then dial out the destination PSTN number.
Most implementations require this setting to be configured to 1.
Min. Delay before
Dial PSTN
Default is 500ms. This needs to be equal to or greater than the Current Disconnect threshold
setting. Once the threshold is reached the gateway can dial out. This parameter should only be
used if there are PSTN line detection issues.
Unconditional Call
Forward to VOIP:
This is an extremely important setting to make sure incoming PSTN calls are picked up and
forwarded to the correct VOIP destination.
User ID - This parameter allows users to configure a User ID or extension number to be
automatically dialed upon FXO line off-hook.
SIP Server - You also need to specify the Profile of the user id configured above (p1 stands for
Profile 1, p2 stands for Profile 2 and so on).
SIP Destination Port - Along with the user-id and Profile, you also have the option to choose
the destination port where you would like to send the call. By default it should be set to ch1-
x:5060; (x can be 4 or 8 depending on number of ports).
We can also specify a different destination for each port. For example under User ID we can
type in: ch1:104;ch2:227;ch3-5:501;ch6,7:856.
Under Sip Server we can type in: ch1:p1;ch2-4:p2;ch5:p3
Under Sip Destination Port we can type in: ch1-2:5060;ch2:7080;ch3-8:5066++
Number of Rings
Before Pickup
Default is 4. This is the number of rings the gateway will wait to send the call to the
VOIP side in case the Caller ID has yet to be detected. If there's CID information the
call will be sent right away. If your lines don't have the CID service set this to 1.
