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Daniel, Thank you! 
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<div class="">You are right about this.
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<div class="">I configured PJSIP not to check whether the contact contains SIPS. </div>
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<div class="">This solved the problem on one of my setups where I have one NIC that has a public IP.</div>
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<div class="">However on the original setup, the kamailio has one public IP and one private IP. In that setup, the ACK to the 200 OK is not forwarded over the private IP to the freeswitch. This only happens in TLS, when I work with TCP it works well. I believe
 it is somehow connected to the record route, and I’m looking into PJSIP to try to find the answer, but is there anything I could do in the kamailio? </div>
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<div class="">I have the same problems with other SIP clients(Bria for example)</div>
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<div class="">Thanks,</div>
<div class="">Arik Halperin<br class="">
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<div class="">On 11 Jun 2018, at 9:43, Daniel-Constantin Mierla <<a href="mailto:miconda@gmail.com" class="">miconda@gmail.com</a>> wrote:</div>
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<p class="">Hello,</p>
<p class="">Kamailio is not involved in the issue reported here. Practically, pjsip expects sips: scheme in the contact URI, which was set by FreeSwitch in 200ok. Maybe there is an option that you have to turn on for FreeSwitch to use sips: scheme.</p>
<p class="">Otherwise, you can try to replace sip with sips in kamailio config and do the reverse the other way.</p>
<p class="">Cheers,<br class="">
Daniel<br class="">
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<div class="moz-cite-prefix">On 05.06.18 06:56, Arik Halperin wrote:<br class="">
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Hello,
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<div class="">I’m using TLS</div>
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<div class="">After receiving 200OK from kamailio:</div>
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<div class="">r2voip.clear2voipdialer I/(NativeSdk_2_0) 1528174138320 PJSIP: (NativeSdk_2_0) 1528174138320 PJSIP:2018-05 07:48:58.319   pjsua_core.c RX 2203 bytes Response msg 200/INVITE/cseq=8107 (rdata0x7a2c56fb38) from TLS 70.36.25.65:443:</div>
<div class="">                                                                                                               SIP/2.0 200 OK</div>
<div class="">                                                                                                               Via: SIP/2.0/TLS 10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias</div>
<div class="">                                                                                                               Record-Route: <sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes></div>
<div class="">                                                                                                               Record-Route: <sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes></div>
<div class="">                                                                                                               From: "number" <sips:<a href="mailto:972523391991@kamprod.telemessage.com" class="" moz-do-not-send="true">972523391991@XXXXXXX.com</a>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO</div>
<div class="">                                                                                                               To: <sips:<a href="mailto:1111111@kamprod.telemessage.com" class="" moz-do-not-send="true">1111111@XXXXXX.com</a>>;tag=64H63g861ajHj</div>
<div class="">                                                                                                               Call-ID: Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2</div>
<div class="">                                                                                                               CSeq: 8107 INVITE</div>
<div class="">                                                                                                               Contact: <<a href="sip:1111111@10.168.10.200:5080;transport=tls" class="" moz-do-not-send="true">sip:1111111@10.168.10.200:5080;transport=tls</a>></div>
<div class="">                                                                                                               User-Agent: FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit</div>
<div class="">                                                                                                               Accept: application/sdp</div>
<div class="">                                                                                                               Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY</div>
<div class="">                                                                                                               Require: timer</div>
<div class="">                                                                                                               Supported: ti</div>
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<div class=""><b class="">PJSIP responds with:</b></div>
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<div class=""><b class="">Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session</b></div>
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<div class="">What is the reason for this? How can I fix this issue?</div>
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<div class="">Thanks,</div>
<div class="">Arik Halperin</div>
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<pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla -- <a class="moz-txt-link-abbreviated" href="http://www.asipto.com/">www.asipto.com</a>
<a class="moz-txt-link-abbreviated" href="http://www.twitter.com/miconda">www.twitter.com/miconda</a> -- <a class="moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda">www.linkedin.com/in/miconda</a>
Kamailio World Conference -- <a class="moz-txt-link-abbreviated" href="http://www.kamailioworld.com/">www.kamailioworld.com</a></pre>
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