<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:verdana, helvetica, sans-serif;font-size:10pt"><div>Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay.</div><div><br></div><div>Is there any other values that I should look at? </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> 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> 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>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 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><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><div></div>
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