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<p style="margin-top:0;margin-bottom:0">Hi. <br>
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<p style="margin-top:0;margin-bottom:0">Just as curiosity, what would cause an ACK lost in your system?</p>
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<p style="margin-top:0;margin-bottom:0">Best regards.<br>
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<div class="PlainText">RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
Ph: +55 35 3471 9200 RAMAL 979<br>
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<div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt" color="#000000" face="Calibri, sans-serif"><b>De:</b> Users <users-bounces@lists.opensips.org> em nome de Ben Newlin <Ben.Newlin@genesys.com><br>
<b>Enviado:</b> terça-feira, 27 de março de 2018 11:18<br>
<b>Para:</b> OpenSIPS users mailling list<br>
<b>Assunto:</b> Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?</font>
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<p class="x_MsoNormal">Rodrigo,</p>
<p class="x_MsoNormal"> </p>
<p class="x_MsoNormal">Yes, they do. I am using them to do exactly what you describe. The final reply (fr) timer is how long a transaction will wait to receive a final reply (>=200). If the timer expires without receiving a final reply the transaction will
be canceled and failure route will be triggered with, I think, a local 408 response.</p>
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<p class="x_MsoNormal">As for $DLG_timeout, you can set that value multiple times in a call. We do this as well. Prior to the call being ACKed we set this value fairly low (~5s) in order to disconnect the dialog if the ACK is lost. Once we receive the ACK,
we then extend it to a much longer value.</p>
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<p class="x_MsoNormal"><span style="color:black">Thanks,</span></p>
<p class="x_MsoNormal"><span style="color:black">Ben Newlin</span></p>
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<p class="x_MsoNormal"><b><span style="font-size:12.0pt; color:black">From: </span>
</b><span style="font-size:12.0pt; color:black">Users <users-bounces@lists.opensips.org> on behalf of Rodrigo Pimenta Carvalho <pimenta@inatel.br><br>
<b>Reply-To: </b>OpenSIPS users mailling list <users@lists.opensips.org><br>
<b>Date: </b>Tuesday, March 27, 2018 at 10:08 AM<br>
<b>To: </b>OpenSIPS users mailling list <users@lists.opensips.org><br>
<b>Subject: </b>Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?</span></p>
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<p><a name="x__MailOriginalBody" id="LPlnk999377" previewremoved="true"><span style="font-size:12.0pt; color:black">Ok Ben.</span></a></p>
<p><span style=""><span style="font-size:12.0pt; color:black"> </span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black">I will check this possibility and see if reply times will change the duration of a not answered call.</span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black">Thank you.</span></span></p>
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<p class="x_MsoNormal"><span style=""><span style="font-size:10.0pt; color:black">RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
Ph: +55 35 3471 9200 RAMAL 979</span></span></p>
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<p class="x_MsoNormal"><span style=""><b><span style="color:black">De:</span></b></span><span style=""><span style="color:black"> Users <users-bounces@lists.opensips.org> em nome de Ben Newlin <Ben.Newlin@genesys.com><br>
<b>Enviado:</b> terça-feira, 27 de março de 2018 10:43<br>
<b>Para:</b> OpenSIPS users mailling list<br>
<b>Assunto:</b> Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?</span></span><span style=""><span style="font-size:12.0pt; color:black">
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<p class="x_xmsonormal"><span style=""><span style="font-size:12.0pt; color:black">You can also use the reply timers in TM to do this:
</span></span><a href="http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout" id="LPlnk985315" previewremoved="true"><span style=""><span style="font-size:12.0pt">http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout</span></span><span style=""></span></a><span style=""><span style="font-size:12.0pt; color:black"></span></span></p>
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<p class="x_MsoNormal" style="margin-top:15.0pt"><span style=""></span><a href="http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout" target="_blank" id="LPlnk772885" previewremoved="true"><span style=""><span style="font-size:16.0pt; font-family:"Segoe UI"; text-decoration:none">tm
Module - openSIPS</span></span><span style=""></span></a><span style=""><span style="font-size:16.0pt; font-family:"Segoe UI"; color:#0078D7"></span></span></p>
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<p class="x_MsoNormal" style="margin-top:15.0pt; line-height:10.5pt"><span style=""><span style="font-size:10.5pt; font-family:"Segoe UI"; color:#666666">www.opensips.org</span></span></p>
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<p class="x_MsoNormal" style="margin-top:15.0pt; line-height:15.0pt"><span style=""><span style="font-size:10.5pt; font-family:"Segoe UI"; color:#666666">TM module enables stateful processing of SIP transactions. The main use of stateful logic, which is costly
in terms of memory and CPU, is some services inherently ...</span></span></p>
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<p class="x_xmsonormal"><span style=""><span style="font-size:12.0pt; color:black">Thanks,</span></span></p>
<p class="x_xmsonormal"><span style=""><span style="font-size:12.0pt; color:black">Ben Newlin</span></span></p>
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<p class="x_xmsonormal"><span style=""><b><span style="font-size:12.0pt; color:black">From:
</span></b></span><span style=""><span style="font-size:12.0pt; color:black">Users <users-bounces@lists.opensips.org> on behalf of Laszlo <laszlo@voipfreak.net><br>
<b>Reply-To: </b>OpenSIPS users mailling list <users@lists.opensips.org><br>
<b>Date: </b>Tuesday, March 27, 2018 at 9:40 AM<br>
<b>To: </b>OpenSIPS users mailling list <users@lists.opensips.org><br>
<b>Subject: </b>Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?</span></span></p>
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<p class="x_xmsonormal"><span style=""><a name="x_x__MailOriginalBody" id="LPlnk185660" previewremoved="true"><span style="font-size:12.0pt; color:black"> </span></a></span><span style=""><span style="font-size:12.0pt; color:black"></span></span></p>
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<p class="x_xmsonormal"><span style=""><span style="font-size:12.0pt; color:black">On Tue, Mar 27, 2018 at 3:31 PM, Rodrigo Pimenta Carvalho <</span></span><a href="mailto:pimenta@inatel.br" target="_blank" id="LPlnk74749" previewremoved="true"><span style=""><span style="font-size:12.0pt">pimenta@inatel.br</span></span><span style=""></span></a><span style=""><span style="font-size:12.0pt; color:black">>
wrote:</span></span></p>
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<p><span style=""><span style="font-size:12.0pt; color:black">Hi.</span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black"> </span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black">When a peer invites another one to a call, there are calling and ringing tones for these peers.</span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black">My SIP agents let these tones execute during 2 minutes. After this, the call is terminated, if no one answers the call.</span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black"> </span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black">How to configure OpenSIPS, if possible, so that any call will be terminated after 1 minute?
</span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black"> </span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black">Any hint will be very helpful!</span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:black">Best regards.</span></span></p>
<p><span style=""><span style="font-size:12.0pt; color:#888888"> </span></span><span style=""><span style="font-size:12.0pt; color:black"></span></span></p>
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<p class="x_xmsonormal"><span style=""><span style="font-size:10.0pt; color:#888888">RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
Ph: +55 35 3471 9200 RAMAL 979</span></span><span style=""><span style="font-size:12.0pt; color:black"></span></span></p>
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<p class="x_xmsonormal"><span style=""><span style="font-size:12.0pt; color:black">Maybe you can play with $DLG_timeout, see </span></span><a href="http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#timeout-pvar-id" id="LPlnk743806" previewremoved="true"><span style=""><span style="font-size:12.0pt">http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#timeout-pvar-id</span></span><span style=""></span></a><span style=""></span><span style="font-size:12.0pt; color:black"></span></p>
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