Setting The Dtmf-Reminder For Voip; Defining Recall Key Functions For Voip (Hook Flash); Defining Local Communication Ports For Voip - Siemens Gigaset S685 IP User Manual

Web configurator –configuring phone via PC
Setting the DTMF-reminder for
VoIP
DTMF signalling is required, for example,
for querying and controlling certain net-
work mailboxes via digit codes or for
remote operation of the integrated
answer machine.
To send DTMF signals via VoIP you must
first define how key codes should be con-
verted into and sent as DTMF signals: as
audible information via the speech chan-
nel or as "SIP Info" message.
Ask your VoIP provider which type of
DTMF transmission it supports.
¤
Open the following Web page:
¢ ¢
Telephony
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.
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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).
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.
122
Settings
Advanced Settings.
area,
2833, if DTMF sig-
RFC
¤
Open the following Web page:
¢ ¢
Telephony
¤
Enter the data you received from your
VoIP provider into the
and
Application Signal
Flash (R-key) area.
¤
Now click to save your settings.
Set
The setting for the recall key applies to all
registered handsets.
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 Yes option if you do not want
the phone to use fixed ports for
and port, but rather to use any free
RTP
ports.
The use of random ports makes sense
if you want several phones to be oper-
ated on the same router with NAT.
Settings
Advanced Settings.
Application Type
fields in the
Hook
Settings
Advanced Settings.
area,
field. If you
SIP port
SIP port