[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=243311#comment-243311 ] 

Harish.K edited comment on ASTERISK-27839 at 5/4/18 8:36 AM:
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Asterisk was installed from this repository https://build.opensuse.org/package/show/network%3Atelephony/asterisk .
Build logs are available here https://build.opensuse.org/public/build/network:telephony/openSUSE_Factory/x86_64/asterisk/_log

>From build logs, I think  libpjsip-2.7.2 is used.

Repository of  libpjsip can be found here https://build.opensuse.org/package/show/network%3Atelephony/libpjsip
Build logs of libpjsip is available here https://build.opensuse.org/public/build/network:telephony/openSUSE_Factory/x86_64/libpjsip/_log


was (Author: harish2704):
Asterisk was installed from this repository https://build.opensuse.org/package/show/network%3Atelephony/asterisk .
Build logs are available here https://build.opensuse.org/public/build/network:telephony/openSUSE_Factory/x86_64/asterisk/_log

>From build logs, I think  libpjsip-2.7.2 is used.

Repository of  libpjsip can be found here https://build.opensuse.org/package/show/network%3Atelephony/libpjsip
Build logs libpjsip is available here https://build.opensuse.org/public/build/network:telephony/openSUSE_Factory/x86_64/libpjsip/_log

> 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



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