Hello everybody<br><br>I have an asterisk with an integration of alcatel pbx, by sip trunk, all calls are fine, but tha calls calls that originate from a analog line, <br><div style="text-align: left;" id="result_box" dir="ltr">
the recipient is not listening, and that if they hear the call originates, the lines are E1 in alcatel pbx.<br><br>When a asteris user call to analog line the call is ok.<br><br><br>Everyone, has been that problem?<br><br>
I change asterisk version 1.4.21 to 1.4.18 but the same problem.<br><br>I saw the cli<br><br>[Aug 12 16:15:40] WARNING[2997]: chan_sip.c:3927 sip_indicate: Don't know how to indicate condition 9<br>[Aug 12 16:15:40] WARNING[2997]: channel.c:2369 ast_indicate_data: Unable to handle indication 9 for 'SIP/4001-0a16f5c0'<br>
<br>Anyone can help me..<br><br><br>Regards<br></div>