#QUICKPICK INVITE CODE PASSWORD#
A password is automatically generated, but you can edit it here. If Create New User is selected above, this will be the auto-generated user's new password. To customize the username, check the Use Custom Username box and enter a username. If you leave the Username field is blank (grayed-out), the username will be the same as the extension number. If Create New User is selected above, this will be the new user's login username. Existing User: If you have any existing users in User Manager who are not already linked to an extension, you can select one here in order to link the user to this extension.None: The extension will not be associated with a user in User Manager.Create New User: Creates a new user in User Manager that will be linked to this extension.An extension may only be linked to one user, and a user may only be linked to one extension. Select a user that this extension should be linked to in the User Management module. A color-coded bar will display the strength of the secret, ranging from "really weak" to "strong." User Manager Settings Link to a Default User A secret is auto-generated but you may edit it. Should be alphanumeric with at least 2 letters and numbers to keep secure. Password (secret) configured for the device. If you leave it blank, the system will use the route or trunk Caller ID, if set. Leave this field blank to disable the outbound CallerID feature for this user. Any setting here will override the common outbound CallerID set in the Trunks module. Format: "caller name" Overrides the CallerID when dialing out a trunk. This will become the Caller ID Name. Only enter the name, NOT the number. This is the name associated with this extension and can be edited any time. We recommend using 3- or 4-digit extension numbers. This will be the extension number associated with this user and cannot be changed once saved. From the Extensions landing page click on the Add New Chan_SIP Extension button.Configuration of these settings is covered by their respective user guides. Options will vary based on installed modules.In Asterisk 12+ there is a PJSIP option which is not covered on this page.
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