Grandstream Networks GXW410X Wait for Dial­tone, Stage Method, Min. Delay before, Dial Pstn

Wait for Dial­tone	Default is No. When set to Yes, the gateway will recognize dial­tone from the Central Office (CO)
	before it completes a call. If you can’t make an outbound call, set this is Yes. if this is set to Yes
	make sure you have configured the dial tone settings correctly in the Channels tab and that there
	is not any major noise interference in the line.
Stage Method	Syntax ­ ch1­8:1; {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 dial­tone 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 off­hook.
	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 user­id 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 ch1­x: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;ch3­5:501;ch6,7:856.
	Under Sip Server we can type in: ch1:p1;ch2­4:p2;ch5:p3
	Under Sip Destination Port we can type in: ch1­2:5060;ch2:7080;ch3­8:5066++
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.