[asterisk-bugs] [JIRA] (ASTERISK-27094) Asterisk 13.15.1 deadlock in fax_gateway_framehook

[David Brillert (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27094?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=237631#comment-237631 ] 

David Brillert commented on ASTERISK-27094:
-------------------------------------------

I'm no expert on gdb traces but I think its locking somewhere in res_fax and I assume in fax_gateway_framehook

The setting which triggers the lock is in sip.conf
setvar          =  FAXOPT(gateway)=yes

Disabling the setting avoids the deadlock but disables all t38 gateway faxing over the SIP trunk.
setvar          =  FAXOPT(gateway)=no

Thread 6 (Thread 0x7fe132c32700 (LWP 2745)):
#0  0x00007fe1e4c26334 in __lll_lock_wait () from /lib64/libpthread.so.0
#1  0x00007fe1e4c215f3 in _L_lock_892 () from /lib64/libpthread.so.0
#2  0x00007fe1e4c214d7 in pthread_mutex_lock () from /lib64/libpthread.so.0
#3  0x000000000053b851 in __ast_pthread_mutex_lock ()
#4  0x000000000045d200 in __ao2_lock ()
#5  0x00000000004688aa in _ast_bridge_lock ()
#6  0x0000000000472cc8 in ast_bridge_peer ()
#7  0x00000000004ca87b in ast_channel_bridge_peer ()
#8  0x00000000004efd25 in ast_unreal_queryoption ()
#9  0x00000000004c2214 in ast_channel_queryoption ()
#10 0x00007fe1c1536cc0 in ast_channel_get_t38_state (chan=0x7fe1c400e5a0)
    at /dar/build/asterisk-13.15.1/include/asterisk/channel.h:2599
#11 0x00007fe1c154d92c in fax_gateway_framehook (chan=0x7fe16c0d0260, f=0x7fe16c062510,
    event=AST_FRAMEHOOK_EVENT_WRITE, data=0x7fe16c0906b0) at res_fax.c:3395
#12 0x0000000000525a6a in framehook_list_push_event ()


> Asterisk 13.15.1 deadlock in fax_gateway_framehook
> --------------------------------------------------
>
>                 Key: ASTERISK-27094
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27094
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_fax
>    Affects Versions: 13.15.1
>            Reporter: David Brillert
>         Attachments: gdbthreadapplyallbt.txt
>
>
> All calling via SIP PSTN carrier.
> progressinband = yes
> directmedia =  yes
> prematuremedia  =  no
> Incoming call A is answered with progress
> Then bridged with progress to external call B
> Call is processed with audio OK
> But no further SIP processing in console and all SIP further signalling dies including OPTIONS packets.
> This looks like a deadlock but asterisk not compiled with DEBUG_THREADS so I attached gdb to asterisk pid and have attached a txt file.



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