[asterisk-bugs] [JIRA] (ASTERISK-27094) res_fax: Deadlock when using Local channels and fax gateway

[nappsoft (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27094?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243518#comment-243518 ] 

nappsoft commented on ASTERISK-27094:
-------------------------------------

Just wanted to inform you that we have a similar problem on one of our systems (about 870 pjsip users, about 359 registered phones, sometimes around 30 concurrent calls). We use chan_pjsip and not chan_sip. Unfortunatelly I can only tell you that we are facing the problem but have no possibility to get more informations at the moment as this is a production system and we have a local sip-client that is automatically restarting asterisk when a deadlock is detected. (So I usually can only look at the system when we already got the alert from our error-recovery tool and when asterisk has already been restarted). => I've started to make sip-traces to confirm that it was not a false alert. It seems like asterisk would still send out "new" packets (outgoing Options for example) but not respond to any incoming traffic (like REGISTER or INVITE).

> res_fax: Deadlock when using Local channels and fax gateway
> -----------------------------------------------------------
>
>                 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, 13.17.0
>            Reporter: David Brillert
>              Labels: fax, pjsip
>         Attachments: core-asterisk-running-2018-05-16T10-28-36-0400.rar, coreshowlocks.txt, deadlock_2018_05_09.rar, gdbthreadapplyallbtfull(1).txt, 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.



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