[Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP

[Emrah]

Actually, your sofia_contact is not happy.. I'm not seeing the transport param for tls in there.
How many profiles are you running?
Are you sure sofia_contact isn't giving you the value of another endpoint registered with UDP?
Also, I went through your message fast, but I don't think you're securing your rtp... 

> On Nov 18, 2016, at 11:08 AM, Tim Smith <randomdev4 at gmail.com> wrote:
> 
> Debian GNU/Linux 8 (jessie)
> Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19)
> x86_64 GNU/Linux
> FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit)
> 
> I have a Vtech handset with TLS/SRTP enabled registered with
> Freeswitch as below:
> 
> 
> Call-ID:        a0000a0a000aa000
> User:           2001 at my.example.com
> Contact:        "my" <sips:2001 at 198.51.100.81:58348>
> Agent:          Vtech Vesa VSP736A 2.0.3.2-0
> Status:         Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646)
> Ping-Status:    Reachable
> Ping-Time:      0.00
> Host:           my
> IP:             198.51.100.81
> Port:           58348
> Auth-User:      2001
> Auth-Realm:     my.example.com
> MWI-Account:    2001 at my.example.com
> 
> 
> sofia_contact is happy :
> 
> freeswitch at my>sofia_contact internal/2001
> sofia/internal/sip:2001 at 198.51.100.81:58348
> 
> I have an inbound dial plan configured as follows:
> 
> <include>
>  <extension name=“test_inbound">
>    <condition field="destination_number" expression=“^(15550100)$">
>     <action application="set" data="domain_name=$${domain}"/>
>      <action application="bridge" data="${sofia_contact(internal/2001)}"/>
>    </condition>
>  </extension>
> </include>
> 
> The problem is Freeswitch is sending invites over SIP/RTP and not
> TLS/SRTP and so the calls never get through :
> 
> INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0
>   Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK
>   From: "Anonymous" <sip:anonymous at 203.0.113.4>;tag=rKmXQjZN8SFXp
>   To: <sip:2001 at 198.51.100.81:58348>
>   m=audio 32190 RTP/AVP 8 98 9 101
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:98 G726-32/8000
>   a=rtpmap:9 G722/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=ptime:20
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org