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</o:shapelayout></xml><![endif]--></head><body lang=NL link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hello,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Just looking at the capture file, it shows that the Kamailio sends a 200 ok on your invite.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The call never gets relayed to the Asterisk box.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Maybe you can take <a href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a> as a guide, get this working and then start modifying to your specific wishes.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Ive learned a lot by using the Asipto how to’s and changing them to my needs.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Rgds.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] <b>On Behalf Of </b>Ravindra Gowda<br><b>Sent:</b> maandag 24 juni 2013 12:21<br><b>To:</b> users@lists.kamailio.org<br><b>Cc:</b> Prabhu R.S<br><b>Subject:</b> [SR-Users] Kamailio+Asterisk Integrated server: 481 dialog/transaction does not exists error<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><div><div><div><div><div><div><div><div><div><p class=MsoNormal style='margin-bottom:12.0pt'>Dear All,<o:p></o:p></p></div><p class=MsoNormal>I have integrated Kamailio (Version 4.0.1) with Asterisk (Version 11.4.0 rc1) server. I should have get VoIP features(call, SMS) with this integrated server. But when i tried to call other end SIP client through this integrated server, I'm getting this '<b>481 dialog/transaction does not exists</b>' error (please find the attachment 'call.pcap' below).<br>SMS is through with the kamailio server without any problem (Find the attachment 'SMS.pcap' below).<o:p></o:p></p></div><p class=MsoNormal>But when it comes to Calling feature, something is wrong and i am unable to find it.<o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'>So please help me in resolving this issue.<o:p></o:p></p></div><p class=MsoNormal>My system configuration is as follows:<o:p></o:p></p></div><p class=MsoNormal> Host Architecture: i686<o:p></o:p></p></div><p class=MsoNormal> OS : Ubuntu 12.04 LTS<o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'> Clients: Wi-Fi enabled VoIP phones.<o:p></o:p></p></div><p class=MsoNormal>Also find the attachments below for the kamailio configuration file and Asterisk's sip.conf and extension.conf files. And tcpdump based trace files for Calling and SMSing process, to give a more complete picture.<o:p></o:p></p><div><div><div><div><div><div><div><div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Any help will greatly appreciate.<br><br clear=all><o:p></o:p></p></div><div><div><div><div><div><p class=MsoNormal><b><span style='color:black'>Thank you and Best regards,</span></b><span style='color:black'><o:p></o:p></span></p></div><div><p class=MsoNormal><b><span style='color:#3333FF'>R Ravindra Gowda</span></b><o:p></o:p></p></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></body></html>