Defining Recall Key Functions For Voip (Hook Flash); Defining Local Communication Ports For Voip - Siemens Gigaset S675IP User Manual

In the
DTMF over VoIP connections
make the required settings for sending
DTMF signals.
¤
Activate or
Audio
nals are to be transmitted acoustically
(in voice packets).
¤
Activate if DTMF signals are to
SIP Info
be transmitted as code.
¤
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).
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.
¤
Open the following Web page:
¢ ¢
Telephony
¤
Enter the data you received from your
VoIP provider into the
and Application Signal
area.
Flash
¤
Now click Set to save your settings.
The setting for the recall key applies to all
registered handsets.
area,
2833, if DTMF sig-
RFC
Settings
Advanced Settings.
Application Type
fields in the Hook
Defining local communication
ports for VoIP
¤
Open the following Web page:
¢ ¢
Telephony
In the Listen ports for VoIP connections
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
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 option if you do not want
Yes
the phone to use fixed ports for
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
¤
Now click to save your settings.
Set
Web configurator
Settings
Advanced Settings.
area,
SIP port field. If you
SIP port
and
SIP port RTP port.
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