Also, for now i think the 4th time, may i suggest you join our irc channel so others may help me in assisting you?<br>I have been working fairly hard to keep up with your constant stream of nearly realtime email requests.<br>
<br><div class="gmail_quote">On Tue, Apr 8, 2008 at 10:57 AM, Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
yes do you have both a pcap and a console trace of the call.<br><br>start freeswitch with TPORT_LOG=1<br><br>TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch<br><br>set debug level<br><br>> console loglevel debug<br><br>
<br>capture all the text on the console <br><br><br>my guess without seeing it is that user 22 is behind nat or a proxy or something and you do not have keepalive turned on so the path for the phone on 22 to send the bye to FS is gone so FS is not getting the bye from phone 22<br>
<br>We would never say 481 because if FS thinks the call leg should still be up it will last until the call is terminated one way or another. Remember, each leg of the bridged call is a separate sip call as we are a b2bua.<div>
<div></div><div class="Wj3C7c"><br>
<br><br><div class="gmail_quote">On Tue, Apr 8, 2008 at 10:25 AM, kokoska rokoska <<a href="mailto:kokoska.rokoska@post.cz" target="_blank">kokoska.rokoska@post.cz</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
Hi all,<br>
<br>
I'm afraid I bug in FreeSWITCH SIP call handling.<br>
The scenario is as follows:<br>
<br>
Very simple dialplan:<br>
<br>
<extension name="E1"><br>
<condition field="destination_number" expression="^(23)$"><br>
<action application="set" data="continue_on_fail=true"/><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="bridge" data="sofia/default/$1%$${domain}"/><br>
<action application="respond" data="181 Call is being forwarded"/><br>
<action application="export" data="sip_h_Diversion=23@$$<br>
{domain};reason=unavailable"/><br>
<action application="transfer" data="22"/><br>
</condition><br>
</extension><br>
<extension name="E2"><br>
<condition field="destination_number" expression="^(22)$"><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="bridge" data="sofia/default/$1%$${domain}"/><br>
</condition><br>
</extension><br>
<br>
<br>
User status:<br>
<br>
Users 21 and 22 are registered, user 22 not.<br>
<br>
<br>
Call flow:<br>
<br>
User 21 calls number 23, recieves back 181 and phone 22 starts ringing<br>
(Diversion header is properly appended).<br>
Than user 22 answers (sends 200 OK and recives ACK) and after short<br>
conversation hangs up.<br>
And user 21's call leg isn't hanged up by Freeswitch (BYE is not sent).<br>
After few seconds user 21 hangs up manualy and recieves 200 OK instead<br>
of 481 Call Leg/Transaction Does Not Exist...<br>
<br>
Is it a bug or am I doing something wrong?<br>
<br>
Best regards,<br>
<br>
kokoska.rokoska<br>
<br>
<br>
PS: If someone interested I have pcap dump of communication.<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><br></div></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>
<a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400
</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400