Hello everybody 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, the recipient is not listening, and that if they hear the call originates, the lines are E1 in alcatel pbx.
When a asteris user call to analog line the call is ok. Everyone, has been that problem? I change asterisk version 1.4.21 to 1.4.18 but the same problem. I saw the cli [Aug 12 16:15:40] WARNING[2997]: chan_sip.c:3927 sip_indicate: Don't know how to indicate condition 9 [Aug 12 16:15:40] WARNING[2997]: channel.c:2369 ast_indicate_data: Unable to handle indication 9 for 'SIP/4001-0a16f5c0' Anyone can help me.. Regards
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