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Summary:ASTERISK-27839: Asterisk crashes due to segfault on new incoming SIP call
Reporter:Harish.K (harish2704)Labels:pjsip
Date Opened:2018-05-04 08:11:36Date Closed:2018-05-04 10:31:01
Priority:MajorRegression?
Status:Closed/CompleteComponents:Resources/res_rtp_asterisk
Versions:15.2.2 Frequency of
Occurrence
Constant
Related
Issues:
Environment:Attachments:( 0) asterisk-backtrace.txt
Description: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
Comments:By: Asterisk Team (asteriskteam) 2018-05-04 08:11:37.882-0500

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].

By: Harish.K (harish2704) 2018-05-04 08:14:02.264-0500

gdb backtrace ( full )

By: Joshua C. Colp (jcolp) 2018-05-04 08:16:31.030-0500

How was Asterisk built? How was PJSIP built? Are you using bundled or not?

It appears as though Asterisk was built against one version of PJSIP, but is using a different one.

By: Harish.K (harish2704) 2018-05-04 08:35:43.794-0500

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

By: Harish.K (harish2704) 2018-05-04 10:29:56.114-0500

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.

By: Harish.K (harish2704) 2018-05-04 10:31:01.122-0500

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.