RTPproxy - thats what I suspected. What i'm guessing is that RTPproxy is engaged but somehow the end points receive each other's Direct IPs in SDP...hmm...tell me if your both end points are on the same LAN and server is placed somewhere in internet !!<br>
<br><div class="gmail_quote">On Tue, Jul 3, 2012 at 3:58 PM, Rodrigo Ferreira <span dir="ltr"><<a href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">rodrigo.ferreira@vipway.net.br</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div dir="ltr">
<div style="font-size:12pt;font-family:'Calibri'">
<div>I will take a look at my rtp .. I’m using rtpproxy on my box, I will try to
get some logs today too</div>
<div style="font-size:small;font-style:normal;text-decoration:none;font-family:'Calibri';display:inline;font-weight:normal">
<div style="FONT:10pt tahoma">
<div> </div>
<div style="BACKGROUND:#f5f5f5">
<div><b>From:</b> <a title="govoiper@gmail.com" href="mailto:govoiper@gmail.com" target="_blank">SamyGo</a> </div>
<div><b>Sent:</b> Tuesday, July 03, 2012 4:34 AM</div>
<div><b>To:</b> <a title="aamir_ryu@yahoo.com" href="mailto:aamir_ryu@yahoo.com" target="_blank">aamir chougule</a> ; <a title="users@lists.opensips.org" href="mailto:users@lists.opensips.org" target="_blank">OpenSIPS
users mailling list</a> </div>
<div><b>Subject:</b> Re: [OpenSIPS-Users] Calls being
disconnected</div></div></div>
<div> </div></div>
<div style="font-size:small;font-style:normal;text-decoration:none;font-family:'Calibri';display:inline;font-weight:normal"><div><div class="h5">Besides
logs and any sip traces as Aamir stated it'd be hard to pin point anything. One
of the big reason for automatically dropping an established call specially after
a specific amount of time i.e 30 seconds..thats because of lack of RTPs flowing
through the server and if media-relaying tools are used they will send hangup to
both ends, even though both ends have their RTP flowing
<u>directly</u>.<br><br>Regards,
<div>Sammy</div>
<div>
<div> </div>
<div class="gmail_quote">On Tue, Jul 3, 2012 at 12:21 PM, aamir chougule <span dir="ltr"><<a href="mailto:aamir_ryu@yahoo.com" target="_blank">aamir_ryu@yahoo.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div>
<div style="FONT-FAMILY:times new roman,new york,times,serif;FONT-SIZE:12pt">
<div><span>Hi Rodrigo,</span></div>
<div><br><span></span></div>
<div><span>The only which I know is after the call gets connected i.e. 200OK
from the other end there should always be an ACK for that 200OK, if there is
no ACK for the 200OK then the call will get disconnected after
sometime.</span></div>
<div><br><span></span></div>
<div><span>Can you please give us the logs for the call by tracing it through
ngrep tool because without any sip traces or opensips log we can't tell what
is going on within the box.</span></div>
<div> </div>
<div style="FONT-FAMILY:times new roman,new york,times,serif;COLOR:rgb(3,61,33);FONT-WEIGHT:bold">Regards,<br><br>Aamir
Chougule<span></span></div>
<div style="BACKGROUND-COLOR:transparent;FONT-STYLE:normal;FONT-FAMILY:arial,helvetica,clean,sans-serif;FONT-SIZE:13px"><span><span style="FONT-FAMILY:times new roman,new york,times,serif;COLOR:rgb(3,61,33);FONT-WEIGHT:bold;TEXT-DECORATION:underline">Cell:
09167989111</span><br></span></div>
<div> </div>
<div style="FONT-FAMILY:times new roman,new york,times,serif;FONT-SIZE:12pt">
<div style="FONT-FAMILY:times new roman,new york,times,serif;FONT-SIZE:12pt">
<div dir="ltr"><font face="Arial">
<hr size="1">
<b><span style="FONT-WEIGHT:bold">From:</span></b> Rodrigo Ferreira <<a href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">rodrigo.ferreira@vipway.net.br</a>><br><b><span style="FONT-WEIGHT:bold">To:</span></b> OpenSIPS users mailling list <<a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a>> <br>
<b><span style="FONT-WEIGHT:bold">Sent:</span></b> Monday, 2 July 2012 10:40
PM<br><b><span style="FONT-WEIGHT:bold">Subject:</span></b> [OpenSIPS-Users]
Calls being disconnected<br></font></div>
<div>
<div>
<div> </div>
<div>
<div dir="ltr">
<div dir="ltr">
<div style="FONT-FAMILY:'Calibri';FONT-SIZE:12pt">
<div>Hey guys,</div>
<div> </div>
<div>I’m having problems with calls being “disconnected” and I dont know where
I can start to look at, because on my CDR all those calls ends with a 200OK,
but that isnt true, because you are talking all the suddenly the call
stop.</div>
<div> </div>
<div>Any ideas where I should start looking at?</div>
<div> </div>
<div style="FONT-FAMILY:'Calibri';FONT-SIZE:12pt">Engº Rodrigo
Ferreira<br>Supervisor de Telefonia<br>VIPWay Telecom<br>Tel.: +55 13
4010-1000<br>Cel.: <a href="tel:%2B55%2013%208136-5839" value="+551381365839" target="_blank">+55 13 8136-5839</a></div></div></div></div></div>
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<p align="left" color="#000000">Nenhum vírus encontrado nessa
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2012.0.2179 / Banco de dados de vírus: 2437/5106 - Data de Lançamento:
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