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.
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.