I try to phone to my friend <sip:toto@example.com>, but nothing happens, no ring, nothing at all.
You must verify that linphone uses the network interface that connects you to the internet (or to the network where calls should go). Use the property box, section Network, to select the correct network interface.
In other case, the person you are contacting may be not reachable at the moment...
Linphone seems to connect to the remote sip url, it rings, but when the callee answers, nothing happens and we can't hear each other.
Most people get problems because they don't choose the correct network interface in the property box, section network. For a dialup connection, it should be "ppp0". Note also that the "lo" interface SHOULD ONLY be used for testing with sipomatic. In other cases, it will fail.
First rise up playback and recording level.
If the voice is sometines cutted, you can modify parameter RTP->jitter compensation in the property box to greater values to avoid this. But it increases the delay transmission.
If linphone cannot open the audio device, check if it has the permission to open /dev/dsp, close all programs able to use audio device (xmms, kaiman...).
Use alsa drivers (see http://www.alsa-project.org). Most distributions still use the old oss kernel-official drivers, that have big latency problems and are often buggy. ALSA drivers are much better. Note that you don't need to recompile linphone at all after installing alsa drivers, you even don't have to change for ALSA mode in the prpperty box, section Audio.