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

[Harish.K (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27839?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243313#comment-243313 ] 

Harish.K commented on ASTERISK-27839:
-------------------------------------

Sorry, In a deeper inspection, I found the root cause of this problem. 
This happended because, I was using pjsip library from a non standard repository. I fixed this issue and now it is running fine. 

Sorry for reporting this bug here without re-thinking and Thank you for providing a very useful hint to solve this issue.

> 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
>            Assignee: Unassigned
>              Labels: pjsip
>         Attachments: asterisk-backtrace.txt
>
>
> 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



--
This message was sent by Atlassian JIRA
(v6.2#6252)