[asterisk-bugs] [JIRA] (ASTERISK-26663) Chan_sip deadlock with local channel and audiohooks

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26663?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton closed ASTERISK-26663.
-----------------------------------

    Resolution: Suspended

Please don't open this issue again for 11. It appears we closed your previous nearly identical issue for the same reason. 

If you can reproduce an issue on a branch that is supported for bug fixes then feel free to open a new issue. Please look at the Asterisk versions page that is linked.

> Chan_sip deadlock with local channel and audiohooks
> ---------------------------------------------------
>
>                 Key: ASTERISK-26663
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26663
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.14.0, 11.25.1
>            Reporter: Ruddy G
>            Assignee: Unassigned
>         Attachments: asterisk-11.14.backtrace, asterisk-11.25.backtrace
>
>
> Hi,
> I discovered what creates a deadlock in asterisk and prevent any new sip registration to occur.
> There are two scenarios. 
> Scenario 1: A local channel calls an application that installs audiohooks on it before calling a SIP user. At the end of the call, we have a deadlock.
> Scenario 2: A SIP user calls a PBX which dial a Local channel with an announcement. That local channel dials another SIP user. This last user answers the phone and transfers it before the announcement is completed.
> Scenario 1:
> Create a p.call file wih a local channel
> Channel: Local/51400000000 at my-inbound
> Application: SayAlpha
> Data: 123456789
> Inside [my-inbound] context, have the channel create an audiohook and then dial a SIP trunk.
> At the end of the call, the PBX SIP module is deadlocked.
> No new registration is allowed.
> Here are the relevant threads backtraces:
> Scenario 2:
> Caller A calls from outside and reach context [my-inbound]
> Inside such context, we do dial a local context with announcement:
> Dial(Local/000 at my-local-context,A(myaudio))
> Inside the [my-local-context], simply call SIP user B
> [my-local-context]
> exten => 000,1,Dial(SIP/userb,30)
> So, user B answers the phone. Before his myaudio.gsm announcement is completed, he transfers the call to another location.
> Basically, B just transfer his own local channel instead of SIP/userA
> A deadlock occurs with the same backtraces.



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