<div dir="ltr"><div dir="ltr"><div class="gmail_default" style="font-family:tahoma,sans-serif">Hi,</div><div class="gmail_default" style="font-family:tahoma,sans-serif"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif">I'm a newbie with the SipSimple application and for a project, i want to create simple caller with a rapsberry. Simply, when pushing a button (GPIO) or virtual (Python), init call to an other SIP account registered into FreePBX.</div><div class="gmail_default" style="font-family:tahoma,sans-serif"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif">But when i try, i got this message:</div><div class="gmail_default" style="font-family:tahoma,sans-serif"><br></div><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">pi@raspberrypi:~ $ sip-audio-session 102</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Using account <a href="mailto:101@192.168.1.247">101@192.168.1.247</a></span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Available audio input devices: None, system_default, Playback/recording through the PulseAudio sound server, USB Audio Device, Digital (S/PDIF), USB Audio Device, USB Audio</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Available audio output devices: None, system_default, Playback/recording through the PulseAudio sound server, USB Audio Device, Digital (S/PDIF), USB Audio Device, USB Audio</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Using audio input device: Playback/recording through the PulseAudio sound server (system default device)</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Using audio output device: Playback/recording through the PulseAudio sound server (system default device)</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Using audio alert device: Playback/recording through the PulseAudio sound server (system default device)</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Available control keys:</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">s: toggle SIP trace on the console</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">j: toggle PJSIP trace on the console</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">n: toggle notifications trace on the console</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">p: toggle printing RTP statistics on the console</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">h: hang-up the active session</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">r: toggle audio recording</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">m: mute the microphone</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">i: change audio input device</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">o: change audio output device</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">a: change audio alert device</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">SPACE: hold/unhold</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Ctrl-d: quit the program</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">?: display this help message</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Initiating SIP audio session from '<a href="mailto:sip%3A101@192.168.1.247">sip:101@192.168.1.247</a>' to '<a href="mailto:sip%3A102@192.168.1.247">sip:102@192.168.1.247</a>' via sip:192.168.1.247...</span><br style="outline:0px;color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;font-size:13px">Erreur de segmentation<span class="gmail_default" style="font-family:tahoma,sans-serif">      <<-- Segmentation Fault</span></span></blockquote><div><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif">The receiver ring, but when answer, nobody at phone (It's normal, the app on raspberry as crashed).</div><div class="gmail_default" style="font-family:tahoma,sans-serif"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif">I make test with sip-session, sip-audio-session, but always segmentation fault. I look too, into all different log of system, but no trace of the crash.</div><div class="gmail_default" style="font-family:tahoma,sans-serif"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif">For installing, i used packages from <a href="http://download.ag-projects.com/SipClient/Raspberry/">http://download.ag-projects.com/SipClient/Raspberry/</a></div><div class="gmail_default" style="font-family:tahoma,sans-serif"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif">Had you ever see this error? Or idea for resolve this?</div><div class="gmail_default" style="font-family:tahoma,sans-serif">Thanks a lot and best regards from Belgium :)</div><div><br></div>-- <div><span class="gmail_default">BINET Jean-Marie</span></div><div><span class="gmail_default" style="font-family:tahoma,sans-serif"><a href="mailto:bineje@gmail.com">bineje@gmail.com</a></span><br><div dir="ltr" class="gmail_signature"><div dir="ltr"><div></div></div></div></div></div></div>