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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=ES-MX link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-US style='color:#1F497D'>Thank you for Your response Daniel.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='color:#1F497D'>I am not sure how to do that but I will figure it out.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='color:#1F497D'>Regards<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='color:#1F497D'>Carlos<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=ES style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext;mso-fareast-language:ES-MX'>De:</span></b><span lang=ES style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext;mso-fareast-language:ES-MX'> sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] <b>En nombre de </b>Daniel Grotti<br><b>Enviado el:</b> jueves, 26 de junio de 2014 03:16 a.m.<br><b>Para:</b> sr-users@lists.sip-router.org<br><b>Asunto:</b> Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Hi,<br>a call trace could help you to understand why your server is not receiving response (which one actually ?) form the Cisco.<br>Is that because the Cisco didn't receive your SIP message? Or is it because Cisco replied but the response didn't reach your server ?<br><br>Try to make a sip trace in order to understand that.<br><br>Daniel<br><br><br><br>On 06/25/2014 06:50 PM, Carlos Rangel wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>Hello<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span lang=EN-US>I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11 taking as a guide Daniel’s tutorial <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>.</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US>I just did not create the voicemail tables because voice mail is handled by Freepbx. I installed the system in a separate box for testing and connected to the Freepbx Production server via IAX trunk. </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US>The system is behind a Cisco Firewall and looks like this</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> Remote User Internet Internal network</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US>Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US>I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user is added with kamctl add user password and as long as the extension is created in Freepbx. </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US>The problem that I have is when try to make a call from the remote Ciscos the call is dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be that the server is not receiving responses from the phone</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='text-autospace:none'><span lang=EN-US style='font-size:10.0pt;font-family:"Lucida Console","serif"'>06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission <a href="mailto:000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22">000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22</a> for seqno 102 (Critical Response) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a></span><o:p></o:p></p><p class=MsoNormal style='text-autospace:none'><span lang=EN-US style='font-size:10.0pt;font-family:"Lucida Console","serif"'>Packet timed out after 32001ms with no response</span><o:p></o:p></p><p class=MsoNormal style='text-autospace:none'><span lang=EN-US style='font-size:10.0pt;font-family:"Lucida Console","serif"'>[2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call <a href="mailto:000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22">000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22</a> - no reply to our critical packet (see <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US>Is this something that we can adjust in kamailio or could it be related to the FW configuration?? Sorry but I am very new to kamailio and sip.</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US>Thanks</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US>Carlos</span><o:p></o:p></p><p class=MsoNormal><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif";mso-fareast-language:ES-MX'><br><br><br><o:p></o:p></span></p><pre>_______________________________________________<o:p></o:p></pre><pre>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<o:p></o:p></pre><pre><a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><o:p></o:p></pre><pre><a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></pre></blockquote><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif";mso-fareast-language:ES-MX'><o:p> </o:p></span></p></div></body></html>