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Defining local communication ports for voip – Siemens GIGASET C475 IP User Manual

Page 148

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147

Web configurator – configuring the telephone via a PC

Defining recall key functions for VoIP (hook flash)

Gigaset C470-475 IP / EN for IM-Ost / A31008-xxxx-xxxx-x-xxxx / web_server.fm / 18.12.07

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Defining recall key functions for VoIP (hook flash)

Your VoIP provider may support special performance features. To make use of these fea-

tures, 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:

Settings

¢

Telephony

¢

Advanced Settings

.

¤

Enter the data you received from your VoIP provider into the fields

Application Type

and

Application Signal

in the

Hook Flash (R-key)

area.

¤

Now click

Set

to save your settings.

The setting for the recall key applies to all registered handsets.

Defining local communication ports for VoIP

¤

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 sig-

nalling 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 low-

est even number will be selected automatically (e.g. you enter 5003, then 5002 is set

automatically). The default port number for voice transmission is 5004.

Use random ports

Click

Yes

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 operated 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

.

¤

Now click

Set

to save your settings.

Please note:

The settings for DTMF signalling apply to all VoIP connections (VoIP accounts).

This manual is related to the following products: