Siemens Gigaset C450 IP User Manual
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Web configurator
Gigaset C450 IP / Greek eng / A31008-M1713-T151-2-8U19 / web_server.fm / 19.9.06
Ve
rs
ion 4,
16
.09.
2005
Username
Enter the caller ID for your VoIP pro-
vider account. This ID is usually identi-
cal to the first part of your SIP address
(URI, your Internet phone number).
Example: If your SIP address is
"[email protected]", enter
"987654321" in
Username
.
Domain
Specify the last part of your SIP address
(URI) here.
Example: For the SIP address
"[email protected]", enter
"provider.com" in
Domain
.
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 1024 and 49152. 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 1024 and 49152. 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
transmission is 5004.
Note:
Ports 0 to 1023 should not be used,
because these are often used by standard
applications.
Note:
Ports 0 to 1023 should not be used,
because these are often used by standard
applications.