Hi Thufir, The analysis of a SIP Debug depends on what the problem to be solved.
If you experience problems with inbound calls from a SIP trunk or provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or from where it is supposed to come call. Then you make a test call, and look in full log an INVITE message (note that you analize an OPTION message in your mail, but I think that this not help in this case). After the incoming INVITE message from your SIP provider, you can follow the rest of the Asterisk logic and look for the reason why Asterisk is denying that call. Hope this help you. -- GnuPG Key ID: 0x39BCA9D8 https://www.github.com/mefhigoseth ...:::[ God Rulz ! ]:::... -- _____________________________________________________________________ -- 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