[asterisk-bugs] [JIRA] (ASTERISK-27839) Asterisk crashes due to segfault on new incoming SIP call

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27839?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243305#comment-243305 ] 

Asterisk Team commented on ASTERISK-27839:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> Asterisk crashes due to segfault on new incoming SIP call
> ---------------------------------------------------------
>
>                 Key: ASTERISK-27839
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27839
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 15.2.2
>            Reporter: Harish.K
>
> h4. OS version
> Distributor ID:	openSUSE
> Description:	openSUSE Tumbleweed
> Release:	20180420
> h4. Steps to reproduce this bug
> * Install asterisk with default configuration. 
> * Add few SIP accounts 
> {code}
> [myusers]
> type=friend
> context=public
> host=dynamic
> secret=1234
> transport=udp
> disallow=all
> allow=ulaw
> allow=h263
> allow=h264                                                                                                                                                    
> allow=h263p
> allow=vp8
> allow=vp9
> qualify=no
> [3333334001](myusers)
> [3333334002](myusers)
> {code}
> * Register any accounts using a SIP client ( I used Jitsi )
> * initiate a sip call to any number



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