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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Really not much to know.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Just read the packet headers and they
usually tell you what is going on.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I learned by just reading them for a
while.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>404 means not there.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>403 means more of an internal error.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I am going threw the learning curve of the
dialplan that will give a better understanding of what is going on.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>In a nut shell, SIP uses port 5060 (most
of the time) to establish authentication.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>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).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Then, bahda-bing!<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>The call begins.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Everything else is in the dial plan.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Or, you could just go spend hours in the
RFC for the particulars!<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b><span style='font-weight:bold'>On Behalf Of </span></b>Arpit Mehta<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, May 31, 2007 4:44
AM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users] How
to read SIP debug?</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>best thing would be to
read the rfc's rahter than any book ... <br>
read rfc3261 ... u could follow it up with other rfcs ....<o:p></o:p></span></font></p>
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<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 5/31/07, <b><span style='font-weight:bold'>Rizwan
Hisham</span></b> <<a href="mailto:rizwanhasham@gmail.com">rizwanhasham@gmail.com</a>>
wrote:</span></font></span><o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Hi all, <br>
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? <br
clear=all>
<br>
<span class=sg>-- </span><br>
<span class=sg>Rizwan Hisham</span><br>
<span class=sg>Software Engineer</span><br>
<span class=sg>AXVOICE Inc. </span><br>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
<br clear=all>
<br>
-- <br>
Arpit Mehta<br>
Graduate Student<br>
Department of Computer Science<br>
<st1:place w:st="on"><st1:PlaceName w:st="on">Columbia</st1:PlaceName> <st1:PlaceType
w:st="on">University</st1:PlaceType></st1:place><br>
<br>
Tel: 1-646-387-5998 <o:p></o:p></span></font></p>
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