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

[Tim Smith]

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