Hello Carlos, I have the same problem when I try to do a flash hook with dahdi module. Did you resolve your problem?
Thanks in advance, Alex 2009/8/13 Carlos Rojas <[email protected]>: > 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 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
