[asterisk-users] How to read SIP debug?

[BSumrall]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <rizwanhasham at gmail.com> 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. 
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-- 
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University

Tel: 1-646-387-5998 

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