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<p>Hello,</p>
<p>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>Otherwise, you can try to replace sip with sips in kamailio
config and do the reverse the other way.</p>
<p>Cheers,<br>
Daniel<br>
</p>
<br>
<div class="moz-cite-prefix">On 05.06.18 06:56, Arik Halperin wrote:<br>
</div>
<blockquote type="cite"
cite="mid:140EA09C-16A9-4FF2-9370-D03E6AAD2B08@mobilinq.io">
<meta http-equiv="Content-Type" content="text/html; charset=utf-8">
Hello,
<div class=""><br class="">
</div>
<div class="">I’m using TLS</div>
<div class=""><br class="">
</div>
<div class="">After receiving 200OK from kamailio:</div>
<div class=""><br class="">
</div>
<div class="">
<div class="">
<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@kamprod.telemessage.com</a>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO</div>
<div class="">
To: <sips:<a
href="mailto:1111111@kamprod.telemessage.com" class=""
moz-do-not-send="true">1111111@kamprod.telemessage.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>
</div>
<div class=""><br class="">
</div>
<div class=""><br class="">
</div>
<div class=""><b class="">PJSIP responds with:</b></div>
<div class=""><b class=""><br class="">
</b></div>
<div class=""><b class="">Secure dialog requires SIPS scheme in
Contact and Record-Route headers, ending the session</b></div>
</div>
<div class=""><b class=""><br class="">
</b></div>
<div class="">What is the reason for this? How can I fix this
issue?</div>
<div class=""><br class="">
</div>
<div class="">Thanks,</div>
<div class="">Arik Halperin</div>
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<br>
<pre wrap="">_______________________________________________
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</pre>
</blockquote>
<br>
<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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