On 2/4/07, 李君 <[EMAIL PROTECTED]> wrote:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ;
(...)
== Channel 'SIP/111-086497c8' jumping out of macro 'nway-start' Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/112-08641920 and SIP/111-086497c8
I have the exact same problem, but as of yet have found no solution. Anybody? Is this a problem with ast_async_goto(), or with the way it is being used? Adolfo _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
