Sending debug to you directly as attachment. and siptrace, which appeared exactly when SIP fantom call comes is below: freeswitch@lab3> send 1185 bytes to udp/[192.168.5.21]:5062 at 19:10:19.075023: ------------------------------------------------------------------------ INVITE sip:sip3779100@192.168.5.21:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKFgg44amt4Z8BN Route: <sip:sip3779100@192.168.5.21:5062> Max-Forwards: 70 From: "" <sip:0@192.168.100.11>;tag=N68Ftp01QyDgp To: <sip:sip3779100@192.168.5.21:5062> Call-ID: 92598766-fdb7-122e-78bc-bcaec59901b2 CSeq: 12623230 INVITE Contact: <sip:mod_sofia@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@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@192.168.5.21:5062> From: "" <sip:0@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@192.168.5.21:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKFgg44amt4Z8BN Route: <sip:sip3779100@192.168.5.21:5062> Max-Forwards: 70 From: "" <sip:0@192.168.100.11>;tag=N68Ftp01QyDgp To: <sip:sip3779100@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@192.168.5.21:5062>;tag=ae5521c6a6820f1ai2 From: "" <sip:0@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@192.168.5.21:5062>;tag=53489ed715c1d150i2 From: "" <sip:0@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@192.168.5.21:5062> Server: Linksys/SPA942-6.1.3(a) Content-Length: 0 ------------------------------------------------------------------------ 2011/5/21 Anton VG <anton.vazir@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@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@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@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@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@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@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@192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062 &park
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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