[Freeswitch-users] Invite on SIP instead of TLS

[Peter P GMX]

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 at 217.xxx.xxx.186:2651 Setting proxy route to
sofia/internal/sip:723329 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 217.xxx.xxx.186:2651>.
Max-Forwards: 69.
From: "Extension 723321" <sip:723321 at 217.xxx.xxx.190>;tag=HB02U2mHX28yK.
To:
<sip:723329 at 217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da>.
Call-ID: e3b50c2c-b3a0-122c-4491-001e904cc34e.
CSeq: 114620396 INVITE.
Contact: <sip:mod_sofia at 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 at 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.