[Freeswitch-dev] SIP fantom calls, when calling disconnected SIP endpoint. EP called when plugged in. suppose bug?
Steven Ayre
steveayre at gmail.com
Sat May 21 08:18:52 MSD 2011
' console loglevel debug' only controls the log level of mod_console (i.e.
when you start FS in foreground with -c)
fs_cli uses its own command "/log debug"
Connect using fs_cli and run:
> /log debug
> sofia global siptrace on
And you should then get debug output.
-Steve
On 20 May 2011 20:13, Anton VG <anton.vazir at gmail.com> wrote:
> Sending debug to you directly as attachment.
>
> and siptrace, which appeared exactly when SIP fantom call comes is below:
>
> freeswitch at lab3> send 1185 bytes to udp/[192.168.5.21]:5062 at
> 19:10:19.075023:
> ------------------------------------------------------------------------
> INVITE sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKFgg44amt4Z8BN
> Route: <sip:sip3779100 at 192.168.5.21:5062>
> Max-Forwards: 70
> From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
> To: <sip:sip3779100 at 192.168.5.21:5062>
> Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
> CSeq: 12623230 INVITE
> Contact: <sip:mod_sofia at 192.168.100.11:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fccbba5 2011-05-18
> 19-00-42 -0400
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary,
> refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 209
> X-MT-Context: None
> X-FS-Support: update_display
> Remote-Party-ID:
> <sip:0 at 192.168.100.11>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1305894626 1305894627 IN IP4 192.168.100.11
> s=FreeSWITCH
> c=IN IP4 192.168.100.11
> t=0 0
> m=audio 23962 RTP/AVP 0 8 3 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ------------------------------------------------------------------------
> recv 296 bytes from udp/[192.168.5.21]:5062 at 19:10:19.088162:
> ------------------------------------------------------------------------
> SIP/2.0 100 Trying
> To: <sip:sip3779100 at 192.168.5.21:5062>
> From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
> Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
> CSeq: 12623230 INVITE
> Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
> Server: Linksys/SPA942-6.1.3(a)
> Content-Length: 0
>
> ------------------------------------------------------------------------
> send 406 bytes to udp/[192.168.5.21]:5062 at 19:10:19.088319:
> ------------------------------------------------------------------------
> CANCEL sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKFgg44amt4Z8BN
> Route: <sip:sip3779100 at 192.168.5.21:5062>
> Max-Forwards: 70
> From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
> To: <sip:sip3779100 at 192.168.5.21:5062>
> Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
> CSeq: 12623230 CANCEL
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> Content-Length: 0
>
> ------------------------------------------------------------------------
> recv 348 bytes from udp/[192.168.5.21]:5062 at 19:10:19.101833:
> ------------------------------------------------------------------------
> SIP/2.0 481 Call Leg/Transaction Does Not Exist
> To: <sip:sip3779100 at 192.168.5.21:5062>;tag=ae5521c6a6820f1ai2
> From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
> Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
> CSeq: 12623230 CANCEL
> Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
> Server: Linksys/SPA942-6.1.3(a)
> Content-Length: 0
>
> ------------------------------------------------------------------------
> recv 378 bytes from udp/[192.168.5.21]:5062 at 19:10:19.109452:
> ------------------------------------------------------------------------
> SIP/2.0 180 Ringing
> To: <sip:sip3779100 at 192.168.5.21:5062>;tag=53489ed715c1d150i2
> From: "" <sip:0 at 192.168.100.11>;tag=N68Ftp01QyDgp
> Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2
> CSeq: 12623230 INVITE
> Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKFgg44amt4Z8BN
> Contact: "sip3779100" <sip:sip3779100 at 192.168.5.21:5062>
> Server: Linksys/SPA942-6.1.3(a)
> Content-Length: 0
>
> ------------------------------------------------------------------------
>
>
> 2011/5/21 Anton VG <anton.vazir at gmail.com>:
> > Anthony, believe or not: There is no debug log. Only SIPTRACE, but
> > it's given already above.
> > Debug log appears if I do not kill a UUID, if UUID killed, no log
> > (except siptrace)
> >
> > 2011/5/20 Anthony Minessale <anthony.minessale at gmail.com>:
> >> you would need to provide a log.
> >>
> >> sofia global siptrace on
> >> console loglevel debug
> >>
> >> reproduce and attach to pastebin.freeswitch.org
> >>
> >>
> >> On Thu, May 19, 2011 at 4:10 PM, Anton VG <anton.vazir at gmail.com>
> wrote:
> >>> originate_timeout does not play role.
> >>> Reproducibility is somewhere around 100%-30% (sometimes with every
> >>> call, sometimes i have to do that 2-3 times to catch it).
> >>>
> >>> - Switch off the phone,
> >>> - make call,
> >>> - switch it on,
> >>> - wait a little to call appear.
> >>>
> >>> but looks like the phone should be plugged in fast, otherwise this not
> happens.
> >>>
> >>> 2011/5/20 Anton VG <anton.vazir at gmail.com>:
> >>>> I was killing not job uuid, I expressed it wrong. I'm killing call
> >>>> UUID, previously created by create_uuid
> >>>>
> >>>> Except this, the meaning is the same, and phantom calls persits ;)
> >>>>
> >>>> 2011/5/19 Anthony Minessale <anthony.minessale at gmail.com>:
> >>>>> I am trying to understand your explanation but based on what you say
> >>>>> if you think you can call uuid_kill on a job uuid you are totally
> >>>>> wrong.
> >>>>>
> >>>>> If you want to know the uuid before originate is over you would need
> >>>>> to provide it as origination_uuid
> >>>>>
> >>>>> bgapi originate {origination_uuid=cluecon}sofia/foo/foo at bar.com 9999
> >>>>>
> >>>>> now you can do
> >>>>>
> >>>>> uuid_kill cluecon at will
> >>>>>
> >>>>> job uuid is only the uuid the result will be in.
> >>>>>
> >>>>> Please stop assuming everything is a bug.
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> On Thu, May 19, 2011 at 10:38 AM, Anton VG <anton.vazir at gmail.com>
> wrote:
> >>>>>> Adding progress_timeout and leg_progress_timeout did not changed it
> behavior
> >>>>>>
> >>>>>> originate
> {originate_retries=0,originate_timeout=60,progress_timeout=5,leg_timeout=60,origination_caller_id_number=907388397,ringback='%(2000,4000,440.0,480.0)',origination_uuid=f7f2d66d-d5e0-4b77-9fda-bae5556b6c8f}[continue_on_fail=false,leg_progress_timeout=7,leg_a_uuid=9bf89608-dd3f-4b28-9ab5-7f71bdd2fbd1]sofia/internal/sip:sip3779100 at 192.168.5.21:5062
> ;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062
> >>>>>> &park
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> 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/
> >>>>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>>>
> >>>>> AIM: anthm
> >>>>> MSN:anthony_minessale at hotmail.com
> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>>>> IRC: irc.freenode.net #freeswitch
> >>>>>
> >>>>> FreeSWITCH Developer Conference
> >>>>> sip:888 at conference.freeswitch.org
> >>>>> googletalk:conf+888 at conference.freeswitch.org
> >>>>> pstn:+19193869900
> >>>>>
> >>>>> _______________________________________________
> >>>>> 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
> >>>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> 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/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> 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
> >>
> >
>
> _______________________________________________
> 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
>
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