Grandstream Networks GXW410X AC Termination, Impedance, Wait for Dial-tone, Stage Method

AC Termination	Selects the impedance of the analog line connected to the FXO port on the GXW410x. Here is
Impedance	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	Default is 500ms. This needs to be equal to or greater than the Current Disconnect threshold
Dial PSTN	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	This is an extremely important setting to make sure incoming PSTN calls are picked up and
Forward to VOIP:	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	Default is 4. This is the number of rings the gateway will wait to send the call to the
Before Pickup	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.

Grandstream Networks, Inc.	GXW410x User Manual	Page 20 of 35
Firmware Version 1.3.4.13		Last Updated: 3/2012