Grandstream Networks GXW410 User Manual page 21

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
setting. Once the threshold is reached the gateway can dial out. This parameter should only be
Dial PSTN
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
forwarded to the correct VOIP destination.
Forward to VOIP:
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:
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.
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ch1­2:5060;ch2:7080;ch3­8:5066++