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SIP Address
Proxy to Use
Subscribe Policy
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.
Your connection to the Internet is broken.
The person you are calling is not reachable.
My call seems to connect, but I cannot hear anything.
The voice output on both ends sounds strangely clipped.
DTMF does not work.
118 SUSE Linux Enterprise Desktop 10 GNOME User Guide
Enter a valid SIP address for your contact.
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.
Your subscribe policy determines whether your presence or absence can be tracked by others.
There are several reasons why a call could fail:
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.
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.
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.
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.
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.
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