Dear All,

Loving freeswitch so far.
There's one thing I need help with though.

I can't seem to get freeswitch to send BYE packets to my originating call when the hangup happens.
At first I tried this in Lua, but the symptoms are still the same with a barebone dialplan:
        <extension name="bridge_call_test">
                <condition field="destination_number" expression="^103">
                        <action application="answer"/>
                        <action application="sleep" data="5000"/>
                        <action application="hangup"/>
                </condition>
        </extension>

I also turned siptrace on with the commands:
sofia profile internal siptrace on
sofia profile external siptrace on

All I can get though is a SIP/2.0 200 OK at the beginning of the call, but no BYE at the end:
Does anyone have an idea as to what I am missing?

2010-06-21 13:10:20.600232 [INFO] switch_cpp.cpp:1142 username: XXXX, password: XXXXXXXX
2010-06-21 13:10:20.603790 [NOTICE] switch_channel.c:669 New Channel sofia/internal/XXXX@aaaaa [6a2c61cc-4f44-4a33-b6fc-
af9728c2f1d2]
2010-06-21 13:10:20.607511 [INFO] mod_dialplan_xml.c:418 Processing XXXX->103 in context default
2010-06-21 13:10:20.618949 [NOTICE] mod_dptools.c:719 Channel [sofia/internal/XXXX@aaaaa] has been answered
send 1228 bytes to udp/[192.168.202.67]:5060 at 17:10:20.619691:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.202.67:5060;rport=5060;branch=z9hG4bKPjf2d4bf79-38fb-45fe-82bf-13b070750f90
   From: sip:XXXX@aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff
   To: <sip:103@aaaaa>;tag=8rcH55QDQpmDD
   Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a
   CSeq: 24285 INVITE
   Contact: <sip:103@192.168.202.31:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Require: timer
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Session-Expires: 1800;refresher=uac
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 251
   Remote-Party-ID: "103" <sip:103@192.168.202.31>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1277110358 1277110359 IN IP4 192.168.202.31
   s=FreeSWITCH
   c=IN IP4 192.168.202.31
   t=0 0
   m=audio 29862 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   ------------------------------------------------------------------------
recv 386 bytes from udp/[192.168.202.67]:5060 at 17:10:20.628813:
   ------------------------------------------------------------------------
   ACK sip:103@192.168.202.31:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 192.168.202.67:5060;rport;branch=z9hG4bKPj65f1608c-62e2-4ca8-973b-8d8daf8a9a6b
   Max-Forwards: 70
   From: sip:XXXX@aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff
   To: sip:103@aaaaa;tag=8rcH55QDQpmDD
   Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a
   CSeq: 24285 ACK
   Content-Length:  0

   ------------------------------------------------------------------------
2010-06-21 13:10:25.618976 [NOTICE] mod_dptools.c:705 Hangup sofia/internal/XXXX@aaaaa [CS_EXECUTE] [NORMAL_CLEARING]
2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1182 Session 2 (sofia/internal/XXXX@aaaaa) Ended
2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/XXXX@aaaaa [CS_DESTROY]