Gigaset A510IP User Manual page 139

Please note
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The settings for DTMF signalling apply to all VoIP connections (VoIP
accounts).
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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:
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Settings Telephony
You can connect an external call to one of your VoIP connections with a second
external participant (depending on the provider). You do this by establishing an
external consultation call to the second participant and pressing the R key on the
handset once you have registered the second participant. The call is transferred.
You can expand or change the settings for call transfer as follows:
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You can activate call transfer by ending the call. The two external participants
are connected with one another when you press the end call key a on the
handset. Your connections with the participants are terminated.
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You can activate direct call transfer. You can then transfer the call before the sec-
ond participant has answered.
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You can deactivate call transfer with the R key if you want to assign a different
feature to the R key (
Defining R key functions for VoIP (hook flash)
You can specify the function for the R key on the Web page:
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Settings Telephony
Your VoIP provider may support special performance features. To make use of these
features, your phone needs to send a specific signal (datapacket) to the SIP server.
You can assign this "signal" as the R function to the R key of the handsets. Prerequi-
site: The R key is not used for call divert (default setting, see above).
If you press this key during a VoIP call, the signal is sent. This requires that DTMF sig-
nalling 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:
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Settings Telephony
The following communication ports are used for VoIP telephony:
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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.
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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 for voice transmission.
Configuring the phone via the Web configurator
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Advanced VoIP Settings
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"Defining R key functions for VoIP (hook
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Advanced VoIP Settings
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Advanced VoIP Settings
flash)").
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