Personal Settings

Users having an account in the SIP Switch, can control some of its settings on this page. The account is set up by the administrator, who also has the ability to block a users from accessing this page, by checking the BU (Block User access) box.

Here you specify if and how your calls are to be forwarded.

• No: No forwarding is made. Only the normally registered clients will ring. (The “normally registered” is the SIP Phone or PC Soft SIP client with your SIP user name.)
• Forward: The normally registered client(s) will not ring - only the one(s) listed in the next field will ring.
• Parallel: All SIP clients will ring simultaneously, both the normally registered and the ones specified in the next field.
• Sequence: The SIP clients will ring one after another, for 25 seconds each until there is an answer. The normally registered will ring first, followed by the ones specified in the next field in the given order. You can control the ringing order by adding the q-parameter after the SIP user name. q is a value between 0 and 1 and if no value is given, q=0.5 is assumed. SIP clients with the highest q-value, e.g. “john@company.com;q=0.6”, will ring first. The default ring length will be overridden if another time is specified in the Voice mail field. You can also set the ring length in seconds by adding the t-parameter after a user name, e.g. “john@company.com;t=15” will make john ring for 15 seconds.
• Deny: means that you will not accept any calls at all.

Here you specify the SIP URI, IP address or phone number (e.g. peter@company.com martin@192.168.0.35 0857144433) to the SIP PC clients, the SIP phones or the PSTN phones that your calls shall be forwarded to. A phone number will be processed by the Dial Plan. For a SIP phone on the same LAN using an external SIP server, it is recommended that you specify the local IP address (e.g. john@192.168.0.36 instead of john@company.com) to avoid restrictions in the external SIP server. You can enter several receivers separated by space or comma.

Here you select whether, and on which condition, your calls shall be redirected to the voice mail server (set up by the administrator).

This feature is only available if the administrator has turned on the RI (Restrict Incoming) function for you. When turned on, calls to you are only accepted from callers specified below or from callers allowed by the administrator in a common list.

• Accept All Incoming Callers: When this is selected, you do not make any personal restrictions at all.
• Accept Only Authenticated or Specified Callers: When you select this, more detailed options will show up where you can define your restrictions:

Here you can select if calls from users handled by this SIP server should all be allowed, or should only be allowed after authentication. This will override the administrator's setting. Note that it is not difficult for a non-authorized user to assume a local user's SIP address (spoof) and make incoming calls when Authentication is not selected.

Callers with any of the entered User IDs and Passwords are allowed.

Callers with any of the SIP addresses entered here are allowed. You can use wildcards to specify allowed callers. ? represents any single character while * represents a string of characters of any length. * is only allowed first, last and just before or after @ (e.g. *@partner.com).

You can list the SIP addresses (e.g. peter@company.com) of users that you will not accept calls from. You can use wildcards to specify allowed callers. ? represents any single character while * represents a string of characters of any length. * is only allowed first, last and just before or after @ (e.g. sex*@* and *@sex*). Note that it is easy for a caller to change SIP address, so a persistent caller may bypass this hinder. In such case, you can ask the administrator to turn on the RI (Restrict Incoming) function for you, whereby this blacklist function will be replaced by more strict methods.

If it is a FXS port telephone user which logs in to this page, there will, at the top of this page, the Call Control fields show up like they are also present on the Telephone Ports GUI, see description there.