Strange... Got it working now... I can receive incoming call... Changed following parameters in additional_a2billing_sip.conf of the DID to: -
qualify=yes canreinvite=no Cheers, Nitesh Guillermo Salas M. wrote: > On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote: > >> When I call from my cell to the above DID, it hits on the Asterisk and >> I >> see A2Billing trying to call SIP/2486543210, but it fails because >> Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No >> route to destination) ". >> > > I know it, but the error is saying that you don't have one 2486543210 > user registred. > > Show us the output of: > > sip show peers > > Regards, > > _______________________________________________ --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
