I have updated my system from SVN 10003 to 13223. I also have updated all libraries etc. Everything works fine (SIP +TLS) when calling internal numbers (conference). However calling internally registered phones does not work. Here are some facts which do not fit together - phones are sucessfully registered via TLS (ok) - debug log show that phone will be called via TLS (port 5061) (ok) - Invite message however is sent via SIP (port 5060) Please see the part of the logs below. Anybody has a clue what happened here? Best regards Peter Debug Log: ============ 2009-05-05 00:49:38 [DEBUG] sofia_glue.c:1599 sofia_glue_do_invite() sip:723329@217.xxx.xxx.186:2651 Setting proxy route to sofia/internal/sip:723329@217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 2009-05-05 00:49:38 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/sip:723329@217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651) State CONSUME_MEDIA 2009-05-05 00:49:38 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/internal/sip:723329@217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 entering state [calling][0] 2009-05-05 00:49:38 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/internal/sip:723329@217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 entering state [terminated][503] 2009-05-05 00:49:38 [NOTICE] sofia.c:3469 sofia_handle_sip_i_state() Hangup sofia/internal/sip:723329@217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] Phone is registered with TLS ============================ Call-ID: OWE1NTAyNWU0MGE2NjI1OWJhZjM1YWJiYWJjZGYzYTI. User: 723329@sip2.mydomain.de Contact: "723329" Agent: eyeBeam release 1102u stamp 52345 Status: Registered(TLS-NAT)(unknown) EXP(2009-05-05 00:58:03) Host: sip2.mydomain.de IP: 217.xxx.xxx.186 Port: 2651 Auth-User: 723329 Auth-Realm: sip2.mydomain.de SIP message instead of TLS message: ==================== U 217.xxx.xxx.190:5060 -> 217.xxx.xxx.186:2651 INVITE sip:723329@217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190;rport;branch=z9hG4bK2tH444a02mQZc. Route: <sip:723329@217.xxx.xxx.186:2651>. Max-Forwards: 69. From: "Extension 723321" <sip:723321@217.xxx.xxx.190>;tag=HB02U2mHX28yK. To: <sip:723329@217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da>. Call-ID: e3b50c2c-b3a0-122c-4491-001e904cc34e. CSeq: 114620396 INVITE. Contact: <sip:mod_sofia@217.xxx.xxx.190:5061;transport=tls>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13223M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 447. P-Key-Flags: keys="3". Remote-Party-ID: "Extension 723321" <sip:723321@217.xxx.xxx.190>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1314931392159531063 5177685988992248857 IN IP4 217.xxx.xxx.190. s=FreeSWITCH. c=IN IP4 217.xxx.xxx.190. t=0 0. m=audio 12556 RTP/SAVP 8 9 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:w/ubmPAP4I5BA1Gv1ZWZzbJkfst2e4cY7bKedcjA.