Siemens Gigaset S685 IP User Manual page 115

Web configurator –configuring phone via PC
If you do not enter a name, your
VoIP phone number will be dis-
name
played.
Ask your VoIP provider if this feature is
supported.
Area: General Provider Data
If you have downloaded the general set-
tings for the VoIP provider from the Sie-
mens configuration server (page 112),
then the fields in this area will be preset
with the data from the download. Gener-
ally speaking, you do not need to config-
ure any settings in this area.
Domain
Specify the last part of your SIP address
(URI) here.
Example: For the SIP address
"987654321@provider.com", enter
"provider.com" in Domain.
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: mypro-
vider.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.
114
User- Registration refresh time
Enter the time intervals at which the
phone should repeat the registration
with the VoIP server (SIP proxy) (a
request will be sent to establish a ses-
sion). The repeat is required so that the
entry of the phone in the tables of the
SIP proxy is retained and the phone can
therefore be reached. The repeat will
be carried out for all activated VoIP
phone numbers.
The default is 180 seconds.
If you enter 0 seconds, the registration
will not be repeated periodically.
Area: Network
Please note:
If you have downloaded the general settings
for your VoIP provider from the Siemens con-
figuration server (page 112), then some fields
in this area will be preset with the data from
the download (e.g. the settings for the STUN
server and outbound proxy).
If your phone is connected to a router with
NAT (Network Address Translation) and/or
a firewall, you must make some settings in
this area so that your phone can be
reached from the Internet (i.e. can be
addressed).
Through NAT, the IP addresses of subscrib-
ers in the LAN are concealed behind the
public IP address of the router.
For incoming calls
If port forwarding is activated or a DMZ is
set up for the phone on the router, no spe-
cial settings are required for incoming
calls.
If this is not the case, an entry in the NAT
routing table (in the router) is necessary in
order for the phone to be reached. This
entry is created when the phone is regis-
tered with the SIP service. In the interest
of security, this entry is automatically
deleted at certain intervals (session time-
out). The phone must therefore confirm
its registration at certain intervals (see
NAT refresh
entry stays in the routing table.
time, page 115), so that the