Gigaset S850A GO User Manual page 107

Or you can explicitly define the type of DTMF signalling:
•
Audio or RFC 2833, if DTMF signals are to be transmitted acoustically (in voice packets).
•
SIP info, if DTMF signals are to be transmitted as code.
DTMF signals cannot be transmitted in the audio path (Audio) on broadband
connections (the G.722 codec is used).
Configuring call transfer via VoIP
You can change the settings for call transfer in the Call Transfer area on the web page:
¤
Settings Telephony
You can transfer an external call to a VoIP connection to a second external participant by
pressing the Recall key (depending on the provider).
Add/change settings for call transfer:
•
Activate call transfer by ending the call. The two external participants will be connected
when you press the End call key
•
Activate direct call transfer. The call can be transferred before the second participant has
answered.
•
Press the Recall key to deactivate call transfer if you want to assign a different feature to the
Recall key ( "Defining Recall key functions for VoIP (hook flash)").
Defining Recall key functions for VoIP (hook flash)
You can specify the function for the Recall key on the web page:
¤
Settings Telephony
Your 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"
as the Recall function to the Recall key on the handsets. Prerequisite: The Recall key is not used
for call transfer (default setting).
If you press this key during a VoIP call, the signal is sent. This requires that DTMF signalling via
SIP info messages is activated on the phone (see above).
Defining local communication ports for VoIP
The settings for the communication ports are on the web page:
¤
Settings Telephony
The following communication ports are used for Internet telephony:
•
SIP port: The communication port via which the phone receives (SIP) signalling data. The
default standard port number is set to 5060 for SIP signalling.
•
RTP port: Two consecutive RTP ports (consecutive port numbers) are required for each VoIP
connection. Voice data is received via one port and control data via the other. The default
standard port number is set to 5004 - 5020.
This setting only has to be changed if the port numbers are already being used by other
participants in the LAN. You can then specify other fixed port numbers or port number ranges
for the SIP and RTP port.
If several VoIP phones are operated on the same router with NAT, it makes sense to use randomly
selected ports. The phones must then use different ports so that the router's NAT is only able to
Advanced VoIP Settings
.
Advanced VoIP Settings
Advanced VoIP Settings
Web configurator
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