User's Manual

48
Web configurator
Gigaset C450 IP / GBR PTT / A31008-M1713-L151-1-7619 / web_server.fm / 15.5.06
Version 4, 16.09.2005
Display name (optional)
Enter any name that should be shown
in the other party's display when you
call him via the Internet (example:
Anna Sand). All characters in the UTF8
character set (Unicode) are permitted.
This name must not exceed 32 charac-
ters
If you do not enter a name, Username is
displayed.
Ask your VoIP provider if this feature is
supported.
Proxy server address
The SIP proxy is your VoIP provider's
gateway server. Enter the IP address or
the (fully-qualified) DNS name of your
SIP proxy server.
Example: myprovider.com.
Proxy server port
Enter the number of the communica-
tion port that the SIP proxy uses to send
and receive signalling data (SIP port).
Port 5060 is used by most VoIP provid-
ers.
Registrar server
Enter the (fully-qualified) DNS name or
the IP address of the registrar server.
The registrar is needed when the
phone is registered. It assigns the pub-
lic IP address/port number to your SIP
address (Username@Domain) that were
used by the phone at registration. With
most VoIP providers, the registrar
server is identical to the SIP server.
Example: reg.myprovider.com.
Registrar server port
Enter the communication port used in
the registrar. It is mainly port 5060 that
is used.
Area: Listen ports
Specify the phone's local ports for VoIP
telephony here. The ports must not be
used by any other subscriber in the LAN.
SIP port
Specify the local communication port
that the phone should use to send and
receive signalling data. Specify a
number between 1 and 65535. The
default port number for SIP signalling is
5060.
RTP port
Specify the local communication port
that the phone should use to send and
receive voice data. Enter an even
number between 1 and 65535. The
port number must not be the same as
the port number in the SIP port field. If
you enter an odd number, the even
number just below it will be set (e.g. if
you enter 5003, 5002 is set). The
default port number for voice transmis-
sion is 5004.
Use random ports
Click on the option Yes, if you do not
want the phone to use fixed ports for
SIP port and RTP port, but rather to use
any free ports.
The use of random ports makes sense if
you want several phones to be oper-
ated on the same router with NAT. The
phones must then use different ports
so that the router's NAT is only able to
forward incoming calls and voice data
to one (the intended) phone.
If you click on No, the phone will use
the ports specified in SIP port and RTP
port.
Please note:
Ports 0 to 1023 should not be used,
because these are often used by standard
applications.
Please note:
Ports 0 to 1023 should not be used,
because these are often used by standard
applications.