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6.4 Creating 
Your 
Own SIP Domain 
If you do not already own a domain name, you can easily get one at a domain name registrar found on 
the web. Alternatively, you can directly select a sub domain of some predefined domain name at a 
DNS service provider like Tekea or dyndns.org. Tekea, found at www.tekea.com provides special SIP 
support, making it very easy to set up your own SIP domain. 
To make your SIP server accessible, your SIP domain name has to be made available in DNS, e.g. by 
your Internet service provider or by the mentioned DNS service providers. 
If you have a fixed IP address, the domain name can be handled by a regular DNS server, but if you 
have a dynamic IP address (allocated by your Internet service provider by DHCP - an address that 
may vary from time to time) you need to use dynamic DNS.  
You can directly get a dynamic DNS account from Tekea or 
dyndns.org, and should then use the dynamic DNS client in 
your ADSL Internet Gate to keep your DNS entry updated. 
From the main menu of your unit, select Network 
Configuration,
 then Dynamic DNS. After having created a 
dynamic DNS account at any of the listed Dynamic DNS 
providers, you fill in the fields on the Dynamic DNS page.  
Your ADSL Internet Gate will then update your dynamic DNS 
account, whenever your IP address changes. 
6.5 The 
SIP 
Switch 
The SIP Switch is an additional software for your ADSL Internet Gate, that integrates the usage of soft PC 
clients, SIP telephones and ordinary telephones, adds PBX functionality and gives a unique connectivity with the 
old telephone network via common VoIP service provider’s SIP accounts. 
You can learn more about the SIP Switch by clicking “Using SIP: Telephony, Video, Messaging” on 
the login page of the web interface. 
You purchase the SIP Switch software from the SIP Switch page of the ADSL Internet Gate. You can 
also evaluate the SIP Switch for free for 10 days. The links for purchase and evaluate are found 
beside the Dial Plan on the SIP Switch page. 
Key functions of the SIP Switch are: 
• 
Convenient dialling between SIP phones, soft SIP clients and ordinary analog telephones, using 
URLs, E.164 numbers and internal “extension” numbers. 
• 
Unlimited PSTN connectivity (in and out) through a simple VoIP provider account! 
• 
Flexible Dial Plan (Setup wizard and downloadable examples to ease setup). 
• 
Flexible and extensive ENUM lookup. 
• 
Internal “extension” numbers, dial 0 (or 9) for outside. 
• 
User accounts and User’s control page. 
• 
Incoming regular phone calls are directly forwarded to the right user. 
• 
Direct mapping to users of incoming PSTN calls. 
• 
Forwarding, forking in parallel and sequence. 
• 
Routing to Voice Mail servers requiring specifics in the request-URI. 
• 
Restriction of incoming callers (Blacklist and/or Allow-list based on various criteria). 
• 
Fallback functionality.