Hi to everybody, <br><br>I have an spa 3102 where i connected an analog phone (in the fxs port) and the pstn line (in the fxo port).<br><br>This is my problem:<br><br>the incoming call doesn't arrive to asterisk.<br><br>
In the spa web page i configured this dialplane:<br><br>(<:line01@192.168.1.220:5060>)<br><br>where line01 is the context in sip.conf, <a href="http://192.168.1.220">192.168.1.220</a> is the asterisk ip and 5060 is the asterisk sip port.
<br><br>[line01]<br>username = usersipura<br>fromuser = usersipura<br>secret = pwdsipura<br>host = <a href="http://192.168.1.222">192.168.1.222</a><br>fromdomain = <a href="http://192.168.1.222">192.168.1.222</a><br>port = 5061
<br>type = friend<br>dtmfmode = rfc2833<br>context = call_in<br>insecure = very<br><br><br>Why?<br>is the dialplane wrong?<br><br>help me, please.<br><br>Damiano.<br>