The question was:<br><br>Are you doing the packet capture on the actual FS box using tshark or tcpdump?<br><br><br><div class="gmail_quote">On Thu, Dec 17, 2009 at 9:48 AM, DJB <span dir="ltr"><<a href="mailto:djbinter@yahoo.com">djbinter@yahoo.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div style="font-family: verdana,helvetica,sans-serif; font-size: 10pt;"><div>Anthony,</div>
<div> </div>
<div>I have pasted the invite sip trace here: <a href="http://pastebin.freeswitch.org/11536" target="_blank">http://pastebin.freeswitch.org/11536</a></div>
<div>Please advise if you need further info.</div>
<div> </div>
<div>Thank you.<br></div>
<div style="font-family: verdana,helvetica,sans-serif; font-size: 10pt;"><br>
<div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><div class="im"><font face="Tahoma" size="2">
<hr size="1">
<b><span style="font-weight: bold;">From:</span></b> Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>><br><b><span style="font-weight: bold;">To:</span></b> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b><span style="font-weight: bold;">Sent:</span></b> Wed, December 16, 2009 3:42:48 PM<br><b><span style="font-weight: bold;">Subject:</span></b> Re: [Freeswitch-users] SIP Re-invite<br></font><br></div><div><div></div><div class="h5">
that means the invite is not matching the call dialog<br>compare the via tags and call-id etc<br><br><br>
<div class="gmail_quote">On Wed, Dec 16, 2009 at 5:29 PM, DJB <span dir="ltr"><<a href="mailto:djbinter@yahoo.com" rel="nofollow" target="_blank">djbinter@yahoo.com</a>></span> wrote:<br>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;" class="gmail_quote">
<div>
<div style="font-family: verdana,helvetica,sans-serif; font-size: 10pt;">
<div>
<p class="MsoNormal"><span style="font-size: 10pt;">We have a customer that we are sending calls to off the FS and here is the issue: </span></p>
<p class="MsoNormal"><span style="font-size: 10pt;"> </span></p>
<p class="MsoNormal"><span style="font-size: 10pt;">Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine </span></p>
<p class="MsoNormal"><span style="font-size: 10pt;"> </span></p>
<p class="MsoNormal"><span style="font-size: 10pt;">They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine </span></p>
<p class="MsoNormal"><span style="font-size: 10pt;"> </span></p>
<p class="MsoNormal"><span style="font-size: 10pt;">30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. </span></p>
<p class="MsoNormal"><font face="Arial, sans-serif"><br></font></p>
<p class="MsoNormal"><font face="Arial, sans-serif">One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds.</font></p>
<p class="MsoNormal"><font face="Arial, sans-serif"><br></font></p>
<p class="MsoNormal"><font face="Arial, sans-serif">We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.</font></p>
<p class="MsoNormal"><font face="Arial, sans-serif"><br></font></p>
<p class="MsoNormal"><font face="Arial, sans-serif">Thank you very much.</font></p></div>
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