[asterisk-bugs] [JIRA] (ASTERISK-23718) res_pjsip_incoming_blind_request: crash with NULL session channel

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23718?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan updated ASTERISK-23718:
-----------------------------------

    Target Release Version/s: 12.4.0

> res_pjsip_incoming_blind_request: crash with NULL session channel
> -----------------------------------------------------------------
>
>                 Key: ASTERISK-23718
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23718
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Resources/res_pjsip_refer
>    Affects Versions: 12.2.0
>            Reporter: Jonathan Rose
>            Assignee: Kinsey Moore
>      Target Release: 12.4.0
>
>         Attachments: backtrace_pjsip_blind_xfer_crash.txt
>
>
> I've only been able to reproduce this with Digium Phones on account of them offering a certain level of control over calls during transfers. Still, Asterisk shouldn't crash on any SIP transactions it receives.
> extension:
> <something to dial the other phone with>
> 001,1,Wait(20)
> 001,n,Answer()
> steps to reproduce:
> * PJSIP/dphone1 dials PJSIP/dphone2, gets into normal 2 party bridge situation
> * PJSIP/dphone1 uses xfer key to transfer PJSIP/dphone2 to extension 001
> * While the transfer is progressing, press the ^ key on the Digium Phone's directional pad to switch the call from the current transfer to the initial session with PJSIP/dphone2
> * Press the "Resume" softkey
> * Attempt to transfer the call to 001 again. This should cause the crash.
> I'm attaching a backtrace which details the crash. It can be mitigated simply by adding a null check against session->channel and returning 404 (probably anything other than 200 really). I'm not attaching a patch since I think that might be a superficial fix.



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