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SUSE Linux Enterprise Desktop 10 GNOME User Guide
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SIP Address
Enter a valid SIP address for your contact.
Proxy to Use
If needed, enter the proxy to use for this particular connection. In most cases, this would just be
the SIP address of the SIP server you use.
Subscribe Policy
Your subscribe policy determines whether your presence or absence can be tracked by others.
To call any contact from the address book, select this contact with the mouse, click
Select
to make
the address appear in the address field of the main window, and start the call with
Call or Answer
as
usual.
7.6 Troubleshooting
I try to call someone, but fail to establish a connection.
There are several reasons why a call could fail:
Your connection to the Internet is broken.
Because Liphone uses the Internet to relay your calls, make sure that your computer is
properly connected to and configured for the Internet. This can easily be tested by trying
to view a Web page using your browser. If the Internet connection works, the other party
might not be reachable.
The person you are calling is not reachable.
If the other party refused your call, you would not be connected. If Linphone is not
running on the other party's machine while you are calling, you will not be connected. If
the other party's Internet connection is broken, you cannot make the connection.
My call seems to connect, but I cannot hear anything.
First, make sure that your sound device is properly configured. Do this by launching any other
application using sound output, such as a media player. Make sure that Linphone has sufficient
permissions to open this device. Close all other programs using the sound device to avoid
resource conflicts.
If the above checks were successful, but you still fail to hear anything, raise the recording and
playback levels under the
Sound
tab.
The voice output on both ends sounds strangely clipped.
Try to adjust the jitter buffer using
RTP properties
in
Preferences > Network
to compensate for
delayed voice packages. When doing this, be aware that it increases the latency.
DTMF does not work.
You tried to check your voice mail using the DTMF pad, but the connection could not be
established. There are three different protocols used for the transmission of DTMF data, but
only two of these are supported by Linphone (SIP INFO and RTP rfc2833). Check with your
provider whether it supports one of these. The default protocol used by Linphone is rfc2833,
but if that fails you can set the protocol to SIP INFO in
Preferences > Network > Other
. If it
does not work with either of them, DTMF transmission cannot be done using Linphone.
Содержание LINUX ENTERPRISE DESKTOP 10 - GNOME 19-06-2006
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Страница 63: ...Office and Collaboration II novdocx ENU 01 February 2006 63 II Office and Collaboration...
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Страница 133: ...Multimedia IV novdocx ENU 01 February 2006 133 IV Multimedia...
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Страница 153: ...Appendixes V novdocx ENU 01 February 2006 153 V Appendixes...
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