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 <sip%3A103@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]