Really not much to know. Just read the packet headers and they usually tell you what is going on.
I learned by just reading them for a while. 404 means not there. 403 means more of an internal error. I am going threw the learning curve of the dialplan that will give a better understanding of what is going on. In a nut shell, SIP uses port 5060 (most of the time) to establish authentication. If authentication is go "both ways", meaning firewall or other anomaly is not blocking the path (telnet tells all!), then and RTP stream is established somewhere between ports 10000 and 20000 (one way audio will usually indicate firewall problems here as well). Then, bahda-bing! The call begins. Everything else is in the dial plan. Or, you could just go spend hours in the RFC for the particulars! _____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arpit Mehta Sent: Thursday, May 31, 2007 4:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to read SIP debug? best thing would be to read the rfc's rahter than any book ... read rfc3261 ... u could follow it up with other rfcs .... On 5/31/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote: Hi all, i need to study the SIP protocol. can anybody tell me about any ebook which could halp me understand the sip protocol, architecture, and how to read and understand the sip signalling when i use "sip debug" in asterisk? -- Rizwan Hisham Software Engineer AXVOICE Inc. _______________________________________________ --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 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998
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