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 email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037>
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 email: david.villasmil.work@gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ
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