Hi all, I have a new installation with asterisk 1.6.1.6 but I'm unable to receive calls from a SIP trunk:
[Oct 2 14:30:09] NOTICE[21554]: chan_sip.c:18523 handle_request_invite: Call from 'user001' to extension 'user001' rejected because extension not found. Are there any changes from 1.6.0 to 1.6.1 (or there is a bug)? Below my simple configuration: sip.conf register => user001:[email protected]/user001 [user001-sip-in] context=default defaultuser=user001 fromuser=001 fromdomain=sip.xxx.it type=user insecure=port,invite secret=pass001 qualify=yes port=5060 nat=no host=sip.xxx.it canreinvite=no --- extensions.conf [default] exten => user001,1,Noop(Inbound call) Thanks and regards Carlo Dimaggio _______________________________________________ -- 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
