If you don't see the call coming in when you have sip debug enabled, then the call is not making it to your server.
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis Sent: Friday, April 22, 2011 4:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cannot call to my server with SIP Op 22-04-11 18:13, Eric Wieling schreef: > > "sip set debug on" should help I've tried it, but no, nothing... -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
