[Freeswitch-dev] second leg of call isnt hanged up by Freeswitch

[Anthony Minessale]

Also, for now i think the 4th time, may i suggest you join our irc channel
so others may help me in assisting you?
I have been working fairly hard to keep up with your constant stream of
nearly realtime email requests.

On Tue, Apr 8, 2008 at 10:57 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes do you have both a pcap and a console trace of the call.
>
> start freeswitch with TPORT_LOG=1
>
> TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch
>
> set debug level
>
> > console loglevel debug
>
>
> capture all the text on the console
>
>
> 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
>
> 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.
>
>
>
> On Tue, Apr 8, 2008 at 10:25 AM, kokoska rokoska <kokoska.rokoska at post.cz>
> wrote:
>
> >
> > Hi all,
> >
> > I'm afraid I bug in FreeSWITCH SIP call handling.
> > The scenario is as follows:
> >
> > Very simple dialplan:
> >
> > <extension name="E1">
> >    <condition field="destination_number" expression="^(23)$">
> >      <action application="set" data="continue_on_fail=true"/>
> >      <action application="set" data="hangup_after_bridge=true"/>
> >      <action application="bridge" data="sofia/default/$1%$${domain}"/>
> >      <action application="respond" data="181 Call is being forwarded"/>
> >      <action application="export" data="sip_h_Diversion=23@$$
> > {domain};reason=unavailable"/>
> >      <action application="transfer" data="22"/>
> >    </condition>
> > </extension>
> > <extension name="E2">
> >    <condition field="destination_number" expression="^(22)$">
> >      <action application="set" data="hangup_after_bridge=true"/>
> >      <action application="bridge" data="sofia/default/$1%$${domain}"/>
> >    </condition>
> > </extension>
> >
> >
> > User status:
> >
> > Users 21 and 22 are registered, user 22 not.
> >
> >
> > Call flow:
> >
> > User 21 calls number 23, recieves back 181 and phone 22 starts ringing
> > (Diversion header is properly appended).
> > Than user 22 answers (sends 200 OK and recives ACK) and after short
> > conversation hangs up.
> > And user 21's call leg isn't hanged up by Freeswitch (BYE is not sent).
> > After few seconds user 21 hangs up manualy and recieves 200 OK instead
> > of 481 Call Leg/Transaction Does Not Exist...
> >
> > Is it a bug or am I doing something wrong?
> >
> > Best regards,
> >
> > kokoska.rokoska
> >
> >
> > PS: If someone interested I have pcap dump of communication.
> >
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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