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

Reply via email to