On Mon, Feb 15, 2010 at 03:29:55AM +0530, cool dude wrote: > i am getting a problem, when a call is received by an sip extension it > receives the call no problem in that. but if somebody calls again on that no > busy tone is displayed.i think its a signal problem. so plz tell me > > how to have disconnect signals enabled in line.
Call comes from an analog (FXO) line? Can you provide a CLI trace of the call? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users