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Summary:ASTERISK-17477: Deadlock with blind transfer of SIP call
Reporter:andgre (andgre)Labels:
Date Opened:2011-02-25 05:29:27.000-0600Date Closed:2011-02-28 14:25:22.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) backtrace-threads-2011-02-25-11_02_17.txt
( 1) backtrace-threads-2011-02-25-11_40_49.txt
( 2) core-show-locks-2011-02-25-11_02_17.txt
( 3) core-show-locks-2011-02-25-11_40_52.txt
( 4) core-show-threads-2011-02-25-11_02_17.txt
( 5) core-show-threads-2011-02-25-11_40_53.txt
Description:Asterisk deadlocks avg 1 time per week after blind transfer



****** ADDITIONAL INFORMATION ******

[2011-02-25 11:00:18] VERBOSE[19152] app_dial.c:     -- SIP/102-00000025 is ringing
....
[2011-02-25 11:00:23] VERBOSE[19152] app_dial.c:     -- SIP/102-00000025 answered Datacard/modem03-0c8e
....
102 initiates blind transfer to 104
....
[2011-02-25 11:00:31] VERBOSE[19152] res_musiconhold.c:     -- Started music on hold, class 'default', on Datacard/modem03-0c8e
[2011-02-25 11:00:31] VERBOSE[19152] file.c:     -- <SIP/102-00000025> Playing 'pbx-transfer.slin' (language 'ru')
......
[2011-02-25 11:00:33] VERBOSE[19152] res_musiconhold.c:     -- Stopped music on hold on Datacard/modem03-0c8
......
[2011-02-25 11:01:34] NOTICE[28962] chan_sip.c: Disconnecting call 'SIP/102-00000025' for lack of RTP activity in 61 seconds
....

sip show channels:
Peer       User/ANR    Call ID       Format           Hold     Last Message    Expiry
10.104.1.5  102       685d26bd4ca421e  0x3c0004 (ulaw| No   Rx:BYE
10.104.1.40 104       MGYyYmE5NTFmMzE  0x4 (ulaw)      No   Rx:INVITE  

             
2 active SIP dialogs

core show channels
Channel              Location             State   Application(Data)            
Datacard/modem01-e79 89111142558@from-gsm Down    AppDial((Outgoing Line))      
SIP/104-00000027     s@macro-dialout-trun Ring    Dial(Datacard/r1/89111142558,3
SIP/102-00000025     s@macro-dial:1       Up      AppDial((Outgoing Line))


[2011-02-25 11:38:19] VERBOSE[30235] app_dial.c:     -- SIP/104-0000000f is ringing
....
[2011-02-25 11:38:21] VERBOSE[30235] app_dial.c:     -- SIP/104-0000000f answered DAHDI/3-1
....
104 initiates blind transfer to 204
....
[2011-02-25 11:38:32] DTMF[30235] channel.c: DTMF end '#' has duration 78 but want minimum 80, emulating on SIP/104-0000000f
[2011-02-25 11:38:32] DTMF[30235] channel.c: DTMF end emulation of '#' queued on SIP/104-0000000f
[2011-02-25 11:38:32] VERBOSE[30235] res_musiconhold.c:     -- Started music on hold, class 'default', on DAHDI/3-1
[2011-02-25 11:38:32] VERBOSE[30235] file.c:     -- <SIP/104-0000000f> Playing 'pbx-transfer.slin' (language 'ru')
...
[2011-02-25 11:38:35] DTMF[30235] channel.c: DTMF end passthrough '#' on SIP/104-0000000f
[2011-02-25 11:38:35] VERBOSE[30235] res_musiconhold.c:     -- Stopped music on hold on DAHDI/3-1
...
[2011-02-25 11:39:36] NOTICE[23389] chan_sip.c: Disconnecting call 'SIP/104-0000000f' for lack of RTP activity in 61 seconds


core show channels
Channel              Location             State   Application(Data)            
SIP/104-0000000f     s@macro-dial:1       Up      AppDial((Outgoing Line))      
DAHDI/4-1            (None)               Up      AppDial((Outgoing Line))      
SIP/301-00000009     s@macro-dialout-trun Up      Dial(DAHDI/r0/3364585,300,Tt)
3 active channels
2 active calls
21 calls processed

sip show channels:
Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry
212.53.40.40     0028910237       461f76385ec02ea  0x0 (nothing)    No                                
10.104.1.12      301              ZTBjNzVlZmY3OGR  0x4 (ulaw)       No       Tx: ACK                  
10.104.1.12      (None)           ZWRiMzc1YzA4NTM  0x0 (nothing)    No       Rx: REGISTER              
10.104.1.40      104              1d1889fc57940d4  0x4 (ulaw)       No       Tx: ACK
Comments:By: Leif Madsen (lmadsen) 2011-02-28 14:25:22.000-0600

As this appears to be an issue with chan_datacard I'm closing this issue. However you should be able to report this issue on the Asterisk Forge. Thanks!