humm, how can this be? It always works. Something is really weird going on in your opensips configuration.<div><br></div><div>OK, let me prepare an alternative configuration for you and send you in 2-3 hours.</div><div><br>
</div><div>BTW, you can call me Muhammad.</div><div><br></div><div>Thank you.</div><div><br><br><div class="gmail_quote">On Sun, Nov 18, 2012 at 12:34 AM, M. KHaled W. Chehab <span dir="ltr"><<a href="mailto:kchehab@icucall.com" target="_blank">kchehab@icucall.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="EN-US" link="blue" vlink="purple"><div><p>dears,</p>
<p>the mediaproxy timeout didnt.suceed any more ideas?</p>
<p>Regards</p>
<p><font color="#888888"></font></p><p><font color="#888888"><font color="#888888">Sent from my android device.</font></font></p><font color="#888888">
</font><p></p><div><div class="h5">
<br><br>-----Original Message-----<br>From: "M.Khaled W Chehab" <<a href="mailto:kchehab@icucall.com" target="_blank">kchehab@icucall.com</a>><br>To: 'OpenSIPS users mailling list' <<a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a>><br>
Sent: Fri, 16 Nov 2012 5:18 PM<br>Subject: Re: [OpenSIPS-Users] calls have max call duration<br><br></div></div></div><div><div class="h5"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Dears,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks for your kindly reply, and I will try to arrange my opensips configuration file and send it back if the media configuration didn’t solve this problem ,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">BTW I am using a mix between a residential and trunking script ,do you have a config tested having a permission ,routing dialplan module tested for small carrier level ,you can send me a copy of it .<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">My old config mediaproxy was <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">stream_timeout = 90<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">on_hold_timeout = 3540<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">so set it now to <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">stream_timeout = 30<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">on_hold_timeout = 600<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I will monitor it now and inform you about the result , but my question is ,is there a solution using signaling function I can implement ?<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Regards<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [mailto:<a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a>] <b>On Behalf Of </b>Muhammad Shahzad<br>
<b>Sent:</b> Friday, November 16, 2012 5:01 PM<br><b>To:</b> OpenSIPS users mailling list<br><b>Subject:</b> Re: [OpenSIPS-Users] calls have max call duration<u></u><u></u></span></p><p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Your opensips config is pretty messed up and hard to read. So, while i try to read it to find the problem, as a workaround you should set RTP timeouts in your mediaproxy configuration file (i.e. config.ini), e.g.<u></u><u></u></p>
<div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">stream_timeout = 30<u></u><u></u></p><div><p class="MsoNormal">on_hold_timeout = 1800<br><br>This will result in automatic call hangup if there is no media for at least 30 seconds. This will reduce wrong billing problem to 30 seconds, and your customer's balance will not be exhausted completely. Similarly if call is on hold for more then 30 minutes and no media is transferred during this time (not even music on hold) then call will hangup (change this timeout to whatever appropriate for you).<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Thank you.<u></u><u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Fri, Nov 16, 2012 at 2:13 PM, M.Khaled W Chehab <<a href="mailto:kchehab@icucall.com" target="_blank">kchehab@icucall.com</a>> wrote:<u></u><u></u></p>
<div><div><p class="MsoNormal">Dears,<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">How to drop the call when a cancel received from UA and a 200 OK from the trunk in the same second, note back the client cdr will be a canceled call and the trunk cdr will be 1 sec duration but on my cdr its the max daialog timeout duration<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">I am searching for a solution for this since two weeks as I tried all trick I know and all didn’t succeed ,<u></u><u></u></p><p class="MsoNormal">Find attached the scenario I have problem with, which you can find it at wireshark time line (30.877027) the Client with IP ( 74.114.101.170) sends a cancel ,and the opensips version 1.8.2 server with IP 89.207.133.144 send the client 200 canceling and in its terms forward the cancel to the trunk provider with IP 74.114.101.190 ,but at 31.015333 wireshark time line the trunk sends a 200 OK then 487 request terminated and the billing stops from the trunk provider .<u></u><u></u></p>
<p class="MsoNormal">The problem is my opensips with IP 89.207.133.144 didn’t hangup the call and go for the max call duration in the dialog module .<u></u><u></u></p><p class="MsoNormal">-how to let opensips cancel the call immediately and stop it with no billing .<u></u><u></u></p>
<p class="MsoNormal">Please find attached my opensips.cfg configuration file <u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">Regards<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">Khaled Chehab<u></u><u></u></p><p class="MsoNormal">Senior NGN Engineer<u></u><u></u></p><p class="MsoNormal"><img border="0" width="117" height="75" src="cid:570caaf5-e566-47a6-a53b-e4228b93dbfe" alt="Description: icucall"><u></u><u></u></p>
<p class="MsoNormal">Operations Office - Lebanon<u></u><u></u></p><p class="MsoNormal">Office : <a href="tel:%2B961%201%20515155%20ext%20300" target="_blank">+961 1 515155 ext 300</a><u></u><u></u></p><p class="MsoNormal">
Mobile : <a href="tel:%2B961%203%20045212" target="_blank">+961 3 045212</a><u></u><u></u></p><p class="MsoNormal">E-mail : <a href="mailto:kchehab@icucall.com" target="_blank">kchehab@icucall.com</a><u></u><u></u></p>
<p class="MsoNormal">MSN ID :<a href="mailto:khalidchehab@hotmail.com" target="_blank">khalidchehab@hotmail.com</a> <u></u><u></u></p><p class="MsoNormal">Skype : k_chehab <u></u><u></u></p><p class="MsoNormal">Web Site: <a href="http://www.icucall.com" target="_blank">http://www.icucall.com</a><u></u><u></u></p>
<p class="MsoNormal"> <a href="http://www.allohi.com" target="_blank">http://www.allohi.com</a><u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p></div></div><p class="MsoNormal" style="margin-bottom:12.0pt">
<br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a><br><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><u></u><u></u></p>
</div><p class="MsoNormal"><br><br clear="all"><u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal">-- <br>Muhammad Shahzad<br>-----------------------------------<br>CISCO Rich Media Communication Specialist (CRMCS)<br>
CISCO Certified Network Associate (CCNA)<br>Cell: <a href="tel:%2B49%20176%2099%2083%2010%2085" value="+4917699831085" target="_blank">+49 176 99 83 10 85</a><br>MSN: <a href="mailto:shari_786pk@hotmail.com" target="_blank">shari_786pk@hotmail.com</a><br>
Email: <a href="mailto:shaheryarkh@googlemail.com" target="_blank">shaheryarkh@googlemail.com</a><u></u><u></u></p></div></div></div></div></div></div><br>_______________________________________________<br>
Users mailing list<br>
<a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Muhammad Shahzad<br>-----------------------------------<br>CISCO Rich Media Communication Specialist (CRMCS)<br>CISCO Certified Network Associate (CCNA)<br>
Cell: +49 176 99 83 10 85<br>MSN: <a href="mailto:shari_786pk@hotmail.com" target="_blank">shari_786pk@hotmail.com</a><br>Email: <a href="mailto:shaheryarkh@googlemail.com" target="_blank">shaheryarkh@googlemail.com</a><br>
</div>