I fixed it with kamilio like this:

    if(is_method("INVITE") && status=~"200") {
        if( subst('/sip:/sips:/ig') ) {
            xlog("L_ERR","[ONREPLY]: Changed contact [$ct]!\n"); 
        }
    }

On the onreply_route

Thanks

David
ᐧ

Regards,

David Villasmil
email: david.villasmil.work@gmail.com
phone: +34669448337

On Sun, May 14, 2017 at 5:20 AM, Sergey Safarov <s.safarov@gmail.com> wrote:

FS not supports SIPS scheme.  Exist jira ticket about this.


вс, 14 мая 2017, 2:10 David Villasmil <david.villasmil.work@gmail.com>:
hello, thanks for replying.

Yes, i thought about this, but asterisk has a patch to do this already without any changes on kamailoi, and I was looking more in the same direction.

Regards,

David
ᐧ

Regards,

David Villasmil

On Sun, May 14, 2017 at 12:58 AM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
If kamailio is under your control, you can change contact back and forth from it (eg, before forwarding to freeswitch)

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On May 13, 2017 5:01 PM, "David Villasmil" <david.villasmil.work@gmail.com> wrote:
Hello guys.

I've got this setup, where the client is sending as contact "sip:" coming from kamailio proxy in tls ip until kamailio. Kamailio forwards to FS in tcp, but freeswitch is answering with "sip:" and this breaks the client who answers "SIPS Required"... 

Is there a way of specifying the contact should be sips ?

Has anyone encountered this and solved it?

Regards,

David Villasmil
ᐧ

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