<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:verdana, helvetica, sans-serif;font-size:10pt"><div>I am sorry; here is the complete one: <a href="http://pastebin.freeswitch.org/11540">http://pastebin.freeswitch.org/11540</a></div><div><br></div><div>Thank you.</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"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> DJB <djbinter@yahoo.com><br><b><span style="font-weight: bold;">To:</span></b> freeswitch-users@lists.freeswitch.org<br><b><span style="font-weight: bold;">Sent:</span></b> Thu, December 17, 2009 9:35:27 AM<br><b><span style="font-weight: bold;">Subject:</span></b> Re: [Freeswitch-users] SIP Re-invite<br></font><br><div style="font-family:verdana, helvetica,
sans-serif;font-size:10pt;"><div>Anthony/Michael,</div><div><br></div><div>I finally have a complete traces of a call at <a rel="nofollow" target="_blank" href="http://pastebin.freeswitch.org/11539">http://pastebin.freeswitch.org/11539</a></div><div><br></div><div>There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison.</div><div><br></div><div>The missing re-invite in FS is at <span class="Apple-style-span" style="font-family: 'Courier New', Courier, monospace; font-size: 11px; color: rgb(51, 51, 51); "><span class="nu0">2009</span>/<span class="nu0">12</span>/<span class="nu0">17</span> <span class="nu0">17</span>:<span class="nu0">25</span>:<span class="nu0">55.207747 </span></span></div><div><font class="Apple-style-span" color="#333333" face="'Courier New', Courier, monospace" size="3"><span class="Apple-style-span"
style="font-size:11px;"><br></span></font></div><div><font class="Apple-style-span" color="#333333" face="'Courier New', Courier, monospace" size="3"><span class="Apple-style-span" style="font-size:11px;"><br></span></font></div><div><font class="Apple-style-span" color="#333333"><span class="Apple-style-span" style="font-size:small;">Thank you.</span></font></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;"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight:bold;">From:</span></b> Anthony Minessale <anthony.minessale@gmail.com><br><b><span style="font-weight:bold;">To:</span></b> freeswitch-users@lists.freeswitch.org<br><b><span style="font-weight:bold;">Sent:</span></b> Thu, December 17, 2009 7:57:42 AM<br><b><span style="font-weight:bold;">Subject:</span></b> Re: [Freeswitch-users] SIP Re-invite<br></font><br>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 rel="nofollow" ymailto="mailto:djbinter@yahoo.com" target="_blank" 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 rel="nofollow" target="_blank" href="http://pastebin.freeswitch.org/11536">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 rel="nofollow" ymailto="mailto:anthony.minessale@gmail.com" target="_blank" href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>><br><b><span style="font-weight:bold;">To:</span></b> <a rel="nofollow" ymailto="mailto:freeswitch-users@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users@lists.freeswitch.org">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 rel="nofollow" ymailto="mailto:djbinter@yahoo.com" target="_blank" href="mailto:djbinter@yahoo.com">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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