Hello,
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.
Otherwise, you can try to replace sip with sips in kamailio config and do the reverse the other way.
Cheers, Daniel
On 05.06.18 06:56, Arik Halperin wrote:
Hello,
I’m using TLS
After receiving 200OK from kamailio:
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: SIP/2.0 200 OK Via: SIP/2.0/TLS 10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias Record-Route: sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes Record-Route: sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes From: "number" <sips:972523391991@kamprod.telemessage.com mailto:972523391991@kamprod.telemessage.com>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO To: <sips:1111111@kamprod.telemessage.com mailto:1111111@kamprod.telemessage.com>;tag=64H63g861ajHj Call-ID: Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2 CSeq: 8107 INVITE Contact: sip:1111111@10.168.10.200:5080;transport=tls User-Agent: FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: ti
*PJSIP responds with:*
*Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session*
What is the reason for this? How can I fix this issue?
Thanks, Arik Halperin
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