11 Jun
2018
11 Jun
'18
6:43 a.m.
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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