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Bizfon Manual II: Administrator's Guide  
Administrator’s Menus 
Bizfon2000 and Bizfon4000 (SW Version 4.1.x) 
 
77
 
VoIP Carrier Wizard 
The VoIP Carrier Wizard is used to define access codes for available VoIP Carrier accounts which will particularly allow you to reach users over IP-
PSTN providers or to call to the peers registered on the certain SIP servers by dialing simple digit combinations.  
For each configured VoIP carrier, the wizard creates a specific IP-PSTN routing rule in the 
table. Additionally, a virtual extension 
automatically generated in
 
 will be registered on the defined VoIP Carrier’s SIP server. The settings of that extension will be 
used to make calls from Bizfon’s users towards the created VoIP Carrier will be placed. 
VoIP Carrier Wizard – Page 1 provides a following option 
of describing the VoIP carrier: 
 
When predefined carrier is selected in the VoIP Carrier 
drop down list, the SIP Server and Port will be already 
predefined in the next page. Manual selection allows you to 
manually set up the VoIP Carrier settings. 
The  Description field allows you to insert an optional 
description of the VoIP Carrier. 
 
 
 
Fig. II-134: VoIP Carrier Wizard page 1 
 
VoIP Carrier Wizard – Page 2 is used to define VoIP 
Carrier Settings. The page contains following components: 
 
1. VoIP Carrier Common Settings 
The  Account Name text field requires a username for 
authentication on the defined SIP server. 
The  Password text field requires a password for 
authentication on the defined SIP server. 
The  Confirm Password text field requires a password 
confirmation. If the input is not corresponding to the one in 
the  Extension Password field, the error message 
“Incorrect Password confirm” will appear. 
The  SIP Server text field requires an IP address or the 
hostname of the SIP server destination party it is registered 
on. 
The SIP Server Port text field requires the port number of 
the SIP server destination party it is registered on. 
 
Fig. II-135: VoIP Carrier Wizard page 2 
 
2. VoIP Carrier Advanced Settings 
The Use RTP Proxy checkbox is applicable only when a route is used for calls towards a configured VoIP Carrier from a peer located outside the 
Bizfon. When this checkbox is selected, the RTP streams between external users will be routed through Bizfon. When the checkbox is not selected, 
RTP packets will move directly between peers.  
UserID requires an identification parameter to reach the SIP server. It should have been provided by the SIP service provider and can be requested 
only for certain SIP servers. For others, the field should be left empty.  
Send Keep-alive Messages to Proxy enables the SIP registration server accessibility to the verification mechanism. Timeout indicates the timeout 
between two attempts of SIP registration server accessibility verification. If a reply is not received from the primary SIP server within this timeout, the 
secondary SIP server will be contacted. When the primary SIP server recovers, SIP packets will continue to be sent to the server. 
A group of Host address and Port text fields respectively require the host address (IP address or the host name), the port number of the Outbound 
Proxy
, Secondary SIP Server and the Outbound Proxy for the Secondary SIP Server. These settings are provided by the SIP servers’ providers 
and are used by Bizfon to reach the selected SIP servers.