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><br><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 class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:times new roman,new york,times,serif"><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;font-weight:bold;color:rgb(3,61,33)">Regards,<br><br>Aamir Chougule<span></span></div><div style="font-style:normal;font-size:13px;background-color:transparent;font-family:arial,helvetica,clean,sans-serif">
<span><span style="font-family:times new roman,new york,times,serif;font-weight:bold;text-decoration:underline;color:rgb(3,61,33)">Cell: 09167989111</span><br></span></div><div><br></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 class="h5"> <br><div>
<div dir="ltr">
<div dir="ltr">
<div style="font-size:12pt;font-family:'Calibri'">
<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-size:12pt;font-family:'Calibri'">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>
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<br></blockquote></div><br></div>