Dear All,<br><br>Loving freeswitch so far.<br>There's one thing I need help with though.<br><br>I can't seem to get freeswitch to send BYE packets to my originating call when the hangup happens.<br>At first I tried this in Lua, but the symptoms are still the same with a barebone dialplan:<br>
<span style="font-family: courier new,monospace;"> <extension name="bridge_call_test"></span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;"> <condition field="destination_number" expression="^103"></span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"> <action application="answer"/></span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;"> <action application="sleep" data="5000"/></span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"> <action application="hangup"/></span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;"> </condition></span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"> </extension></span><br><br>I also turned siptrace on with the commands:<br><span style="font-family: courier new,monospace;">sofia profile internal siptrace on</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">sofia profile external siptrace on</span><br><br>All I can get though is a SIP/2.0 200 OK at the beginning of the call, but no BYE at the end:<br>Does anyone have an idea as to what I am missing?<br>
<br>2010-06-21 13:10:20.600232 [INFO] switch_cpp.cpp:1142 username: XXXX, password: XXXXXXXX<br>2010-06-21 13:10:20.603790 [NOTICE] switch_channel.c:669 New Channel sofia/internal/XXXX@aaaaa [6a2c61cc-4f44-4a33-b6fc-<div id=":98" class="ii gt">
af9728c2f1d2]<br>
2010-06-21 13:10:20.607511 [INFO] mod_dialplan_xml.c:418 Processing XXXX->103 in context default<br>2010-06-21 13:10:20.618949 [NOTICE] mod_dptools.c:719 Channel [sofia/internal/XXXX@aaaaa] has been answered<br>send 1228 bytes to udp/[192.168.202.67]:5060 at 17:10:20.619691:<br>
------------------------------------------------------------------------<br> SIP/2.0 200 OK<br> Via: SIP/2.0/UDP 192.168.202.67:5060;rport=5060;branch=z9hG4bKPjf2d4bf79-38fb-45fe-82bf-13b070750f90<br> From: sip:XXXX@aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff<br>
To: <sip:103@aaaaa>;tag=8rcH55QDQpmDD<br> Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a<br> CSeq: 24285 INVITE<br> Contact: <sip:103@192.168.202.31:5060;transport=udp><br> User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported<br>
Accept: application/sdp<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br> Require: timer<br> Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer<br>
Session-Expires: 1800;refresher=uac<br> Min-SE: 120<br> Content-Type: application/sdp<br> Content-Disposition: session<br> Content-Length: 251<br> Remote-Party-ID: "103" <<a href="mailto:sip%3A103@192.168.202.31" target="_blank">sip:103@192.168.202.31</a>>;party=calling;privacy=off;screen=no<br>
<br> v=0<br> o=FreeSWITCH 1277110358 1277110359 IN IP4 192.168.202.31<br> s=FreeSWITCH<br> c=IN IP4 192.168.202.31<br> t=0 0<br> m=audio 29862 RTP/AVP 0 101<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br> a=silenceSupp:off - - - -<br> a=ptime:20<br> ------------------------------------------------------------------------<br>recv 386 bytes from udp/[192.168.202.67]:5060 at 17:10:20.628813:<br> ------------------------------------------------------------------------<br>
ACK sip:103@192.168.202.31:5060;transport=udp SIP/2.0<br> Via: SIP/2.0/UDP 192.168.202.67:5060;rport;branch=z9hG4bKPj65f1608c-62e2-4ca8-973b-8d8daf8a9a6b<br> Max-Forwards: 70<br> From: sip:XXXX@aaaaa;tag=64797d89-3747-4240-abe0-28cd75241aff<br>
To: sip:103@aaaaa;tag=8rcH55QDQpmDD<br> Call-ID: 14b9b208-6690-495e-a403-ae7da6ddd07a<br> CSeq: 24285 ACK<br> Content-Length: 0<br><br> ------------------------------------------------------------------------<br>
2010-06-21 13:10:25.618976 [NOTICE] mod_dptools.c:705 Hangup sofia/internal/XXXX@aaaaa [CS_EXECUTE] [NORMAL_CLEARING]<br>2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1182 Session 2 (sofia/internal/XXXX@aaaaa) Ended<br>
2010-06-21 13:10:25.629655 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/XXXX@aaaaa [CS_DESTROY]</div>