[Freeswitch-dev] SIP fantom calls, when calling disconnected SIP endpoint. EP called when plugged in. suppose bug?
While emulating end-user behavior, encountered a problem. When there is a call to the SIP endpoint, which have disappeared (power unplugged), SOFIA queues the call and calls that endpoint, if it's plugged IN back, within origination_timeout interval. So it means, that there is NO job UUID (I kill all job uuid's by UUID_kill, no show calls, no show channels. But still - I do plug my phone back, it registers to FS, and in 10 seconds I'm receiving a call to it! Reproducibility 100% How to reproduce: Register an endpoint. Unplug the power, so registration persists in FS I do 'originate' a call to a SIP endpoint and &park as shown originate {ringback='%(2000,4000,440.0,480.0)',origination_caller_id_number=907388397,originate_timeout=60,origination_uuid=f84558f4-46e9-4f8a-a546-7edfd6fd0d5e,leg_timeout=60}[continue_on_fail=false,leg_a_uuid=bb744c31-92cf-47d1-aaeb-1c984cb4d599]sofia/internal/sip:sip3779100@192.168.5.21:5062;fs_nat=yes;fs_path=sip%3Asip3779100%40192.168.5.21%3A5062 &park than System attempts to call that endpoint, wait for 4 seconds, disconnect. Kill all job UUIDs, which was originating the call. than plug in the phone, and in 10 seconds there is an incoming call. Just discovered: If i do not kill the JOB UUIDs - than, when phantom call happens, I can see the debug: 2011-05-19 19:59:56.662932 [WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia profile 'internal' for [sip3779100@192.168.100.11] from ip 192.168.5.21 2011-05-19 20:00:08.182927 [INFO] sofia.c:748 sofia/internal/sip:sip3779100@192.168.5.21:5062 Update Callee ID to "Outbound Call" <sip3779100> 2011-05-19 20:00:08.252929 [DEBUG] sofia.c:4770 Channel sofia/internal/sip:sip3779100@192.168.5.21:5062 entering state [proceeding][180] 2011-05-19 20:00:08.252929 [NOTICE] sofia.c:4848 Ring-Ready sofia/internal/sip:sip3779100@192.168.5.21:5062! 2011-05-19 20:04:24.822927 [DEBUG] switch_channel.c:2592 (sofia/internal/sip:sip3779100@192.168.5.21:5062) Callstate Change RINGING -> HANGUP 2011-05-19 20:04:24.822927 [NOTICE] mod_commands.c:2117 Hangup sofia/internal/sip:sip3779100@192.168.5.21:5062 [CS_CONSUME_MEDIA] [NORMAL_CLEARING] 2011-05-19 20:04:24.822927 [DEBUG] switch_channel.c:2608 Send signal sofia/internal/sip:sip3779100@192.168.5.21:5062 [KILL] 2011-05-19 20:04:24.822927 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:sip3779100@192.168.5.21:5062 [BREAK] 2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 API cmd result: [+OK ]2011-05-19 20:04:24.822927 [DEBUG] mod_python.c:186 Finished calling python script 2011-05-19 20:04:24.822927 [DEBUG] switch_cpp.cpp:988 sofia/external/907388397@10.34.0.4 destroy/unlink session from object 2011-05-19 20:04:24.822927 [DEBUG] switch_core_state_machine.c:497 Hangup Command with Session python(cleanup f84558f4-46e9-4f8a-a546-7edfd6fd0d5e): If I KILL the uuids by api hangup hook - there is no DEBUG, but still the call... 2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 hangup hook, UUIDS: f84558f4-46e9-4f8a-a546-7edfd6fd0d5e 2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 API cmd: [uuid_setvar f84558f4-46e9-4f8a-a546-7edfd6fd0d5e disposition ORIGINATOR_CANCEL] 2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 API cmd result: [+OK ]2011-05-19 20:04:24.822927 [CONSOLE] switch_cpp.cpp:1197 API cmd: [uuid_kill f84558f4-46e9-4f8a-a546-7edfd6fd0d5e] show version FreeSWITCH Version 1.0.head (git-fccbba5 2011-05-18 19-00-42 -0400) Seems JOB is detached from UUID - but still not killed itself.
Yea, there is siptrace, when enabled. sofia global siptrace on freeswitch@lab3> send 1181 bytes to udp/[192.168.5.21]:5062 at 15:16:40.952027: ------------------------------------------------------------------------ INVITE sip:sip3779100@192.168.5.21:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKN2QvX33eK72rr Route: <sip:sip3779100@192.168.5.21:5062> Max-Forwards: 70 From: "" <sip:907388397@192.168.100.11>;tag=KXXUQ57gS66ZS To: <sip:sip3779100@192.168.5.21:5062> Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2 CSeq: 12573020 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-FS-Support: update_display Remote-Party-ID: <sip:907388397@192.168.100.11>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1305790653 1305790654 IN IP4 192.168.100.11 s=FreeSWITCH c=IN IP4 192.168.100.11 t=0 0 m=audio 27516 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 304 bytes from udp/[192.168.5.21]:5062 at 15:16:40.968185: ------------------------------------------------------------------------ SIP/2.0 100 Trying To: <sip:sip3779100@192.168.5.21:5062> From: "" <sip:907388397@192.168.100.11>;tag=KXXUQ57gS66ZS Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2 CSeq: 12573020 INVITE Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr Server: Linksys/SPA942-6.1.3(a) Content-Length: 0 ------------------------------------------------------------------------ send 414 bytes to udp/[192.168.5.21]:5062 at 15:16:40.968378: ------------------------------------------------------------------------ CANCEL sip:sip3779100@192.168.5.21:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKN2QvX33eK72rr Route: <sip:sip3779100@192.168.5.21:5062> Max-Forwards: 70 From: "" <sip:907388397@192.168.100.11>;tag=KXXUQ57gS66ZS To: <sip:sip3779100@192.168.5.21:5062> Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2 CSeq: 12573020 CANCEL Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ recv 356 bytes from udp/[192.168.5.21]:5062 at 15:16:40.984188: ------------------------------------------------------------------------ SIP/2.0 481 Call Leg/Transaction Does Not Exist To: <sip:sip3779100@192.168.5.21:5062>;tag=1f405cc57aec781ei2 From: "" <sip:907388397@192.168.100.11>;tag=KXXUQ57gS66ZS Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2 CSeq: 12573020 CANCEL Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr Server: Linksys/SPA942-6.1.3(a) Content-Length: 0 ------------------------------------------------------------------------ recv 386 bytes from udp/[192.168.5.21]:5062 at 15:16:40.994546: ------------------------------------------------------------------------ SIP/2.0 180 Ringing To: <sip:sip3779100@192.168.5.21:5062>;tag=4482dc0be7df9455i2 From: "" <sip:907388397@192.168.100.11>;tag=KXXUQ57gS66ZS Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2 CSeq: 12573020 INVITE Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr Contact: "sip3779100" <sip:sip3779100@192.168.5.21:5062> Server: Linksys/SPA942-6.1.3(a) Content-Length: 0
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
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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Also, there's a handy API call to create a UUID so you don't need to reinvent the wheel: http://wiki.freeswitch.org/wiki/Mod_commands#create_uuid -Steve On 19 May 2011 18:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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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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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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
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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
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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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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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
_______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev@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@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
_______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev@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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-- 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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' 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@gmail.com> wrote:
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 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev@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@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
_______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev@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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-- 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
_______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev@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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participants (3)
-
Anthony Minessale
-
Anton VG
-
Steven Ayre