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@my.example.com Contact: "my" <sips:2001@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@my.example.com sofia_contact is happy : freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@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@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@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