Siemens Gigaset C450 IP User Manual

Page 53

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52

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.

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