Help for SIP Switch

Copyright Intertex Data AB 2005
The SIP Switch Software Addition for the
Intertex Internet Gate (IX66 and IX67)
Overview
Key functions of the SIP Switch are:
• Convenient dialling between SIP phones, soft SIP clients and PSTN, using URLs, E.164
numbers and internal “extension” numbers.
• Flexible Dial Plan (View and download examples!)
• ENUM lookup
• Internal “extension” numbers, dial 0 (or 9) for outside
• User accounts and User’s control page
• Direct mapping to users of incoming PSTN calls
• Forwarding, forking in parallel and sequence
• Voice mail routing
• Restriction of incoming callers (Blacklist and/or Allow-list based on various criteria)
Noteworthy functions are:
• Unlimited PSTN connectivity (in and out) through a simple VoIP SIP account!
• Routing to Voice Mail servers requiring specifics in the request-URI
• Fallback functionality
• Control over who is allowed to use the SIP Switch services
• Extensive Dial Plan wizard to ease setup
And these features are available without limiting the general SIP functionality – so it is not
“just POTS replication”! And only when required, is the Internet Gate acting as a B2BUA,
e.g. when using another user’s account for PSTN connectivity. Whenever possible, the
Internet Gate still acts as a proper SIP proxy.
Also see the short PowerPoint presentation “TheSIPSwith.pps” for a quick overview of
the basics, reasons behind and features of the SIP Switch!
PSTN Connectivity for Everyone
Having your own SIP server may be great, but it is not easy to arrange your own PSTN
connectivity. Available VoIP SIP services, most often offer bi-directional PSTN connectivity
(and sometimes only that). So why not use it for just that?
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With the SIP Switch you can set up the SIP Switch to use a single ordinary VoIP account as
your PSTN gateway for all users! This works with most SIP service provider’s accounts, e.g.
Vonage, and we have not yet seen a limitation in having multiple incoming and outgoing
PSTN calls through one account! (With the Vonage service, you have to subscribe to their
soft client service to get the authentication password. Use the telephone number as the user
name and user ID – domain name will be filled in by the Internet Gate.)
You can even set up the Dial Plan to use different PSTN connections in different countries,
heavily reducing your international traffic cost. You get virtual foreign offices, by routing the
incoming foreign PSTN calls (local in that country) to your local SIP phones.
Please inform us the service provider SIP accounts you are testing against! We like to add
them to the Dial Plan Wizard and other documentation.
Fallback Functionality
The following fallback functions have been implemented:
• In most fields (e.g. dial plan forwards, voice mail server and outbound proxy) you can list
several destinations, which will be tried until one succeeds.
• Each of the Dial Plan, Extension Number and Incoming Call Matching functions can be
set to On, Off or FallBack. Fallback means that the function becomes active if the WAN
connection is down or if the SIP server being routed to is out of order. The SIP Switch
monitors functionality of the SIP servers listed in the “Also apply number processing to
domain(s)” field under the General Settings on the SIP Switch field.
SIP, the SIP Server and the SIP Switch
SIP (Session Initiation Protocol) is an Internet protocol for live communication between
persons. The Internet Gate is equipped with a SIP proxy and registrar that dynamically control
the firewall to allow multiple SIP clients on the LAN to communicate universally over the
Internet.
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In addition, the proxy and registrar can be configured to be a standalone SIP server.
Furthermore, the SIP Switch extensions (Dial Plan, User Accounts and more) integrate SIP
phones with soft PC Clients and ordinary telephones via gateway services.
The SIP Switch is a software option that is purchased by pressing the Buy link above the Dial
Plan table or below the SIP Account table. Without the SIP Switch, only the general settings
plus the SIP user, id and password in the User Account table are functional.
To evaluate the SIP Switch (without having to buy it), press the "Evaluate" link or the
"View/Get Example" link below the Dial Plan table! You then get access to the full SIP
Switch including 5 user accounts for testing purposes, for a maximum period of 10 days or
until this product is restarted.
Note that there are detailed context sensitive help texts available on the SIP Switch page. Just
click the yellow question marks (?)! (Only the English language help texts are updated.)
Start using the SIP Switch – The Quick Guide
Surf into the configuration by entering 192.168.0.1 as the address in your web browser.
Unless otherwise noted, you find the settings in the Configuration-SIP Switch page.
1) Get your SIP phones and/or soft SIP PC clients up and running behind the Internet Gate
Internet Gate (This first step and functionality, for up to 5 users, does not require the
optional SIP Switch.)
a. If your SIP equipment is registered at an operator’s SIP server: No set up is
required! (SIP transparency is pre selected on the Configuration-Security
page
)
b. If you want to use the Internet Gate as your SIP server:
i. Fill in:
Click Save!
ii. Get a DNS entry (Static or Dynamic DNS) to make your SIP domain
globally accessible.
.
2) Specify the SIP domains that the SIP Switch functions shall apply to.
a. If your SIP equipment is registered at an operator’s SIP server, fill in:
b. If you use the Internet Gate as your SIP server: No setup required.
3) Run the Dial Plan Wizard and download a Dial Plan by clicking the
link below the Dial Plan. If you have not purchased the SIP Switch, you will get a full
evaluation copy for 10 days.
a. Make sure the Off/On/FB setting are
for the functions you want to use
(Dial Plan, Extension number, Incoming Call Matching).
b. Modify the Dial Plan according to your needs.
c. If you use a SIP Service provider for PSTN connectivity and want it for all SIP
Switch users, make sure that SIP Account is correctly entered (by the Wizard).
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d. And forward the incoming calls on those special accounts.
e. Click Save!
4) Go to the SIP Account section. If a voice mail server is available, fill in:
.
5) Then for each user:
a. Fill in the user name and password:
and in case you want:
b. Ext. Internal extension number
c.
d.
e.
in case a PSTN gateway directly addresses the user
with a number.
6) On the Internet Gate start page, each user can log in to control his personal settings. The
user shall enter his SIP User Name and Password (as specified in the SIP Account table)
after clicking
.
Happy testing! Report bugs, findings and ideas to us!
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Detailed Description of Each SIP Switch Setting and Function
General SIP Server Settings
Use as SIP server for domain(s) - After checking the box, this product will be the SIP server for the
SIP domains (e.g. "mysipdomain.com myprivat.se") entered, see Setting up Your Own SIP Server.
Authentication Realm - Whenever this SIP server requests authentication, it will present itself with
the realm and ask the client for the user id and password.
Also apply number processing to domain(s) - The number processing by the SIP Switch can be
applied to other domains in addition to the ones for which this product acts as the SIP server.
Typically used to add local functions for clients on the LAN, being registered at an operator's SIP
server.
Allow to Register: - Specifies which SIP clients that are allowed to register on this SIP server.
"Inside users" are clients on the LAN, while "Outside users" are clients outside this firewall. If
Authentication is selected, a user id and a password have to be entered for each user in the SIP
Account table below. Only registered users can receive incoming SIP calls.
Maximum registered SIP addresses - Shows the maximum number of SIP registrations (including
pass through registrations to outside SIP servers) that this product is licensed to handle. Click the
"Buy More" link to get more licenses.
Allow outgoing calls from - Specifies who is allowed to use this SIP proxy for other purposes than
calling the users of this SIP server. "Inside" in All/Inside/None refers to clients on the LAN. Calls
from authenticated users are allowed when checked.
Match full SIP URL for incoming calls
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- If checked, an incoming call will only be passed to the user whose complete SIP URL matches,
otherwise only the user name (the part before @) will be matched. If several users match (when box
not checked), all of them will ring. If full match is used, users who have registered themselves with a
symbolic name (e.g. [email protected]), cannot be reached if the caller instead uses the fixed IP
addresses (e.g. [email protected]) and vice versa.
Read more online:
General SIP Server Settings
Dial Plan
A SIP user may enter a full URL (e.g. [email protected]) or just a number, when wanting
to dial someone. If a number is entered, the Dial Plan will process it and take the appropriate
action - e.g. forward the call to a PSTN gateway - according to rules entered. Running the
Dial Plan wizard at the web page you reach by the "View/Get Example" link below the table,
will create a Dial Plan that you thereafter can modify for specific needs.
The Dial Plan can be turned On, Off or used in fallback (FB) mode. In fallback mode, the
dial plan is inactive unless the WAN connection down or a particular SIP Server to be routed
to is out of order. As a backup, the Dial Plan then becomes active.
The Dial Plan requires the SIP Switch option. Press the "Buy" link above the Dial Plan table
to buy it or press the "Evaluate" link to make the SIP Switch functional for your testing!
Pressing the View/Get example link under the Dial Plan table, takes you to a web site where
you can download examples of Dial Plans or run a wizard that will create your specific Dial
Plan. The wizard is still an early version – please report your findings suggestions!
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The above example is a simple Swedish Dial Plan created by the wizard. The PSTN
connectivity (in and out) is arranged through an ordinary SIP account [email protected] at the
Swedish service provider WX3.
The line below Dial Plan (also created by the wizard) specifies the SIP account that will be
used as PSTN gateway by all users of the SIP Switch. It is entered in the SIP Account table of
the SIP Switch. As indicated by the Account Type, it is not a common User Account in the
SIP Switch, but a special one where the From-header is eXchanged and the SIP Switch
registers itself with the service
.
US Dial Plan Example
Above is an example of a Dial Plan for US, with many rows just to exemplify various
possibilities, emulating a PBX using 9 to get an outside line. The number range 10 – 29 is
reserved for internal “extension” numbers (defined in the SIP Account table below). A dialled
number is compared to the Prefix+Head+Tail columns from top down, until a match is found,
whereby the Action is performed. When the Action is Forward, the Default PSTN Gateway
will be used whenever no other gateway is entered in the Forward to field. Here three
different gateways to the telephone network (PSTN) are used. “localgw” is the single line
gateway built into some models of the Internet Gate running the SIP Switch. It is used for
emergency calls. The wx3 account is used for all calls to Sweden, while other PSTN calls
uses the Vonage account. Explanations to each row in the example above:
1. Emergency number 911 is routed to the local PSTN gateway built into the Internet
Gate.
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2. Emergency number dialled as 0911 (for an outside line) is handled the same way.
3. The 8 prefix is used for allowing anything to pass transparently through the dial plan.
If the called has a numeric SIP user name that may be caught in the dial plan
otherwise, e.g. [email protected] would be trapped at line 9 if not dialled as
[email protected].
4. + is used as with mobile phones, to indicate that a full E.164 number (including
country code) follows. First an Enum lookup is performed to check if a SIP address
exists, in which case the call will be routed directly to the SIP address. If no SIP
address existed, the call will be forwarded to a PSTN gateway (in this case the Default
PSTN Gateway “gatewaypool.com”.
5. * fills the same function as + in the previous row, for SIP phones being unable to enter
the + sign. The A& before Forward in the Action column means that the caller will be
authenticated (checked for UserID and Password as specified in the User Account
table) before the call is routed to the gateway.
6. Calls to Sweden (country code 46) are trapped separately on this line, and forwarded
to the Swedish VoIP provider wx3. Before the call is forwarded to wx3, an Enum
lookup is done to check whether the call can be terminated via SIP instead. For the
Enum lookup, a full E.164 number is required, so in this case the Swedish country
code 46 (in the Add Prefix Enum column) is added before the area code If Enum fails,
the Swedish long distance code 0 is added before the area code and the call is
forwarded to the wx3 account.
7. On this row, the Swedish 071 pay numbers are trapped and will be denied. It must be
done before row 8 to avoid that the 071 numbers are regarded as any number starting
with an area code.
8. Other international calls are trapped on this line. The 9 is for an “outside line”, the
following 011 is the international code used from US. If the Enum lookup returns no
SIP address, the call is routed to Vonage SIP account specified in the Default PSTN
Gateway field (since the Forward to field is left blank).
9. Numbers dialled with 9 for an outside line followed by a 1 for a long distance call are
trapped here. The gateway needs the 1 to know that an area code follows, while Enum
lookup needs the US country 1 to precede the area code, both cases handled by the
Add Prefix columns.
10. Here local numbers (without area code) are trapped. The single 9 just emulates
reaching an outside line. Since the users are located in Dallas, the long distance code
1214 must be added for the gateway and the country plus area code 1214 for the Enum
lookup.
11. Any other number with three digits or more (e.g. as used by some SIP operators) will
pass the Dial Plan unaffected (Allow), and used in the URL as is.
12. Numbers consisting of one or two digits (three are allowed in row 11) are denied here.
(Note that internal “extension” numbers never reaches the Dial Plan.)
Reserved range for internal numbers - A range for internal "extension" numbers can be
entered here. (Each specific internal number is entered in the SIP Account table.) Numbers
within this range will not be processed in the Dial Plan table.
Dial Plan Table - A dialled number is matched against the three first columns in the table,
Prefix, Head and Tail. If a match is found, the Action is performed; else the next row in table
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is checked for a match. The processing order is from top down, until a match is found.
Prefix - The characters in the Prefix+Head+Tail field are compared against the dialled
number. The characters in the Prefix field (e.g. 0 for an outside line or * for some special
service) must be an exact match to the beginning of the dialled number and are stripped off
before an Action is taken.
Head - The Prefix+Head field must also be an exact match to the beginning of the dialled
number.
Tail - The Tail field specifies the characters that are allowed in the remainder of the dialled
number (after an exact match of Prefix+Head).
Minimum Tail - If, after a match in the three previous fields, the length of the tail part is less
than specified in this field, the dialled number is regarded as incomplete.
Action - This field specifies the action(s) that should be performed after a dialled number has
been matched by the previous fields.
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"Allow" means that the SIP URL is used as is.
"Forward" means that the number is sent (as a SIP user name) to the domain specified
in the "Forward to" field (typically used for routing calls for an ordinary telephone, to
a PSTN gateway).
"Enum" results in a request to find whether a SIP address exists for the particular
number. If a SIP URL is found, the call is forwarded to that URL.
"Force" means that the call is forwarded to the domain specified, irrespective of from
where the call originated (e.g. used for emergency calls).
"Deny" means that the call will be rejected (e.g. used for too short numbers).
"&A" before an action instructs this proxy to Authenticate (check user id and
password) before the action is performed.
When two actions are specified, the second is performed if the first fails.
Forward to - Normally, the domain name or IP address of the PSTN gateway to which a call
shall be forwarded to is specified here. However, you can also specify a SIP user that has an
account with a SIP service provider, in which case all users of this product can use that
account as a gateway to the ordinary telephone network! (The correct account type has to be
set up in SIP Account table.) The dialled number, stripped from the Prefix and preceded by
the new Add Prefix Forward in the next column, is passed on as the SIP user name. If there
are several entries (separated by space or comma), the next will be tried if the first fails. If the
Action is Forward, but nothing is specified in this field, the Default PSTN Gateway specified
below this column will be used.
Add Prefix Forward - This prefix will be added in front of the dialled number before the call
is forwarded. It may, e.g. be used for adding a country or an area code before sending the
dialled number to a gateway.
Add Prefix Enum - This prefix will be added in front of the dialled number before an Enum
lookup is done. Typical usage is for adding country code, since Enum uses the full E.164
number.
Enum Root
- In the public Enum service, the root
is ".e164.arpa". However, for local use or testing, other roots may be used. You can also add
several roots (separated by space or comma), that each will be searched until a SIP address is
returned. Last, preceded by an exclamation mark, you may also add a root for searching a
directory with entries that shall NOT be forwarded to a gateway. Thus, if an entry is found in
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e.g. the "!nopstnphone.com" root, the call will NOT be forwarded to a gateway, thus avoiding
the risk for loops. The dialled number, stripped from the Prefix and preceded by Add Prefix
Enum in the previous column (in reverse order and with dots between each digit) will precede
each root searched in the Enum DNS lookup.
Default PSTN Gateway - The domain or IP address entered here will be used if the Action is
forward, but no Forward to address is entered. You may also specify a SIP user that has an
account with a SIP service provider as described above under the "Forward to" field. If more
than one domain is entered (separated by space or comma), the next will be tried if the first
fails.
Since the order in which the rows comes in the Dial Plan table is of importance, there are
radio buttons to mark a particular row, and a Move button to move a row before another
specified row.
Read more online:
Dial Plan
Outbound Proxy
This product acts as an outbound proxy by itself and this table does not have to be filled in.
However, if a user is subscribing to an operator service, the operator may require that all
outbound requests be sent to the operator's proxy. Note that the Dial Plan will forward request
irrespective of this setting.
Send to - Here you enter the domain name or IP address to the SIP proxy that shall receive
the outbound requests.
for Requests from Domain - When the callers SIP URL (SIP address) matches the pattern
specified here, the request will be sent to the proxy in the previous column. You can use
wildcards to specify the callers. ? represents any single character while * represents a string of
characters of any length. * is only allowed first, last and just before or after @. If this field is
left blank, _all_ requests (not trapped by previous entries in this table) will be sent to the
proxy in the previous column.
Thus, if callers are registered at different servers, you can specify the outbound proxy for
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them separately and also set a default outbound proxy.
Read more online:
Outbound Proxy
SIP Accounts and Incoming Call Processing
The SIP Account table below is used for authentication of SIP users and for extra
functionality (columns unlined with pink) that requires the SIP Switch option. Press the "Buy"
link below the SIP Account table to buy the SIP Switch or press the "Evaluate" link to make it
functional for your testing!
Voice mail server - Enter the domain or IP address of the voice mail server here. If more than
one domain is entered (separated by space or comma), the next will be tried if the first fails.
If the voice mail server requires specifics in the request-URI, you can specify it as
exemplified:
o sip:$(cfg.user)@vmserver.com
o sip: vmserver.com;mailbox="$(cfg.user)@$(cfg.host)"
o sip:$(to.user)@$(to.host);maddr= vmserver.com
The variables within $() can be:
- cfg.user = user from SIP Account table
- cfg.host = host from SIP Account table
- ruri.user = user from Request-URI
- ruri.host = host from Request-URI
- to.user = user from To header
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- to.host = host from To header
- from.user = user from From header
- from.host = host from From header
and will be replaced with strings from the configuration or SIP message. "user" and "host"
refers to the user and host parts in a SIP address sip:user@host.
Allow external callers to use internal numbers - If checked, an outside caller can use the
internal "extension" number to call a local user, e.g. by calling [email protected].
Account Type - If the Dial Plan forwards a call to a SIP user specified in the SIP Account
table, it will be processed as specified by the type below. This allows all users of the SIP
Switch to e.g. use a single SIP account with connectivity to the ordinary telephone network!
The full user@host SIP address needs to be specified for the accounts to be used from the
Dial Plan. Incoming calls can be forwarded to any users as specified in the user action
columns.
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"User" Ordinary user account.
"Reg" Register on behalf of client - No SIP phone is connected so the SIP Switch
registers the user on the SIP Server.
"XF" Exchange From header - When used by another client, the From header is
exchanged to the account owner's.
"XF&Reg" Exchange From header and Register - The From header is exchanged and
the SIP Switch registers (both as described above).
"Vng" Vonage - Special type for Vonage SIP accounts with soft PC client. Only
phone number (not full URL) needs to be entered as SIP User name.
"Vng&Reg" Vonage and Register - For Vonage SIP accounts without using a Vonage
client - the SIP Switch registers.
"MR" Media Relay - In addition to the From header exchange, media is always sent
via the SIP Switch.
"MR&Reg" Media Relay and Register - Media is relayed, the From header is
exchanged and the SIP Switch registers.
"Domain" Domain with authentication - The SIP Switch authenticates all users e.g.
towards a gateway at the specified host address.
Ext. = Internal extension number - Here you specify the internal "extension" number you
want to assign to each user. You select any unique number from the reserved range specified
above the Dial Plan table.
SIP User Name or SIP Address (URL) - If "Match full SIP URL for incoming calls" is
selected under the General SIP Server Settings, you should enter the full SIP Address (URL)
(e.g. [email protected]) in this column, otherwise just enter the SIP User Name (e.g.
"peter" of peter@...). This is also the User Name you enter when logging into the personal
page through "User Log in" at the first web page of this product. (For Accounts to be used by
the Dial Plan, the full URL always has to be specified.)
Authentication User ID - This is the User ID used for authentication of SIP requests.
Authentication for registering at this SIP server can be turned on in the General SIP Server
Settings above and authentication may also be forced in the Dial Plan. If authentication is
turned on, the User ID in the SIP client must match the one entered here. The user can change
this field on his personal page.
Authentication Password - This is the Password used for authentication and for logging in to
the personal page. If authentication is turned on, the Password in the SIP client must match
the one entered here. The user can change this field on his personal page.
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Comment - It is useful to enter the user's name here, if he has a numeric SIP name.
Dyn. Regs. - Dynamic Registrations. The number of SIP clients that have been registered for
each user is shown here.
BU = Block User - When checked, this user will not be allowed to log in to his personal page
to change his settings. The personal page is reached through the "User Log in" on the first
web page of this product. You use the SIP name of column 1 and the Password of column 3 to
log in.
RI = Restrict Incoming - By checking this box, the user will only be reachable for callers
defined by the "Allowed Incoming Callers" lists. Also see next point!
AC = Accept Common - There is both a personal list (controlled from the personal page, see
BU above) and a common list (on this page) to specify the Allowed Incoming Callers. If RI is
selected, then this box selects whether the common list on this page will be used in addition to
the user's own list.
Incoming Call Matching - A PSTN gateway may pass an incoming call with the dialled
number in the request-URI. Here you can match one or several incoming PSTN numbers
(separated by space or comma) to specific users. If the account is of "Reg"-type, the SIP
Switch will use the number entered here in the contact header when registering.
Show other dynamically registered SIP users - There may be other SIP users registered,
that you have not created a SIP account for (which is not necessary). You can view these by
checking this box. When viewing these, you can mark and add such users to the SIP Account
table.
Read more online:
SIP Accounts
Incoming Call Processing
Forward Action - Specifies if and how this user's calls are to be forwarded. The user can
change this setting on his personal page. This and the two following columns are only shown
if you have pressed the "Show User Action" button.
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"No" No forwarding is made. Only the Dynamically Registered clients will ring.
"Forward" The Dynamically Registered client(s) will not ring - only the one(s) listed
in the next column will ring.
"Parallel" All SIP clients will ring simultaneously, both the Dynamically Registered
and the ones specified in the next column.
"Sequence" The SIP clients will ring one after another, for 25 seconds each until there
is an answer. The Dynamically Registered will ring first, followed by the ones
specified in the next column 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.
"[email protected];q=0.6", will ring first. The default ring length will be overridden
if another time is specified in the Voice Mail column. You can also set the ring length
in seconds by adding the t-parameter after a user name, e.g.
"[email protected];t=15" will make john ring for 15 seconds.
"Deny" means that this user will not accept any calls at all.
to SIP URL, IP Address, Phone Number or Ext. - Here you specify the SIP PC clients, the
SIP phones or the PSTN phones that the calls for this specific user will be forwarded to. An
entered 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 internal extension
number or the local IP address (e.g. [email protected] instead of [email protected]) to
avoid restrictions in the external SIP server. You can enter several receivers separated by
space or comma. The user can change this field on his personal page.
Voice Mail - Here you select whether, and on which condition, this user's calls shall be
redirected to the Voice mail server (specified above this table).
Incoming Call Blacklist
SPAM is plaguing the efficient use of email. To avoid the same situation for calls over IP,
there may be a need not to accept calls from anyone. Blacklisting is one of the methods
provided. Note that it is easy for a caller to change SIP address, so this method is quite easy to
bypass.
Users with SIP addresses listed here are not allowed make incoming calls or otherwise use
this SIP server. The SIP addresses may include wildcards. ? represents any single character
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while * represents a string of characters of any length. * is only allowed first, last and just
before or after @ (e.g. sex*@* and *@sex*). Each user can also set up his own blacklist in
addition to this one, on his personal page.
Allowed Incoming Callers
SPAM is plaguing the efficient use of email. To avoid the same situation for calls over IP,
there may be a need not to accept calls from anyone. When the RI box in the SIP Account
table is checked, no calls are allowed for that user unless specified in the personal "Allow
Incoming Callers" section or, when AC is checked, unless specified in the common "Allow
Incoming Callers" section here.
Local users - Here you can select if calls from users handled by this SIP server should be
allowed or allowed only after authentication. 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.
Authenticated - Callers with any of the entered User IDs and Passwords are allowed.
Not Authenticated - 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).
Read more online:
Restrict Incoming Callers
Export/Import Settings
You can save the settings on this page as a file on you hard disk by pressing Export. When
Importing settings from a previously stored file, you can select which parts of the settings you
want to restore. Only the General SIP Settings can be restored if you have not purchased the
SIP Switch.
It is recommended that you make a security back-up of your settings by Exporting them
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to a file on your hard disk!
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