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Web configurator –configuring phone via PC
Gigaset S675 IP / ALGERIAN eng / A31008-M1915-A751-1-3T19 / web_server.fm / 27.10.08
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Open the following Web page:
Settings
¢
Telephony
¢
Advanced Settings
.
In the
DTMF over VoIP connections
area,
make the required settings for sending
DTMF signals.
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Activate
Audio
or
RFC 2833
,
if DTMF sig-
nals are to be transmitted acoustically
(in voice packets).
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Activate
SIP Info
if DTMF signals are to
be transmitted as code.
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Now click
Set
to save your settings.
Defining recall key functions for
VoIP (hook flash)
Your VoIP provider may support special
performance features. To make use of
these features, your phone needs to send
a specific signal (data packet) to the SIP
server. You can assign this "signal" to your
phone's recall key.
If you press the recall key during a VoIP call
the signal will be sent to the server.
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Open the following Web page:
Settings
¢
Telephony
¢
Advanced Settings
.
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Enter the data you received from your
VoIP provider into the
Application Type
and
Application Signal
fields in the
Hook
Flash (R-key)
area.
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Now click
Set
to save your settings.
The setting for the recall key applies to all
registered handsets.
Defining local communication
ports for VoIP
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Open the following Web page:
Settings
¢
Telephony
¢
Advanced Settings
.
In the
Listen ports for VoIP connections
area,
specify which local ports the telephone is
to use for VoIP telephony. 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 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 next lowest
even number will be selected automat-
ically (e.g. you enter 5003, then 5002
is set automatically). The default port
number for voice transmission is 5004.
Use random ports
Click the
Yes
option 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
No
, the phone will use the
ports specified in
SIP port
and
RTP port
.
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Now click
Set
to save your settings.
Please note:
– The settings for DTMF signalling apply to all
VoIP connections (VoIP accounts).
– DTMF signals can not be transmitted in the
audio path (
Audio
) on broadband connec-
tions (the G.722 codec is used).