Summary: | ASTERISK-27839: Asterisk crashes due to segfault on new incoming SIP call | ||
Reporter: | Harish.K (harish2704) | Labels: | pjsip |
Date Opened: | 2018-05-04 08:11:36 | Date Closed: | 2018-05-04 10:31:01 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | 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. |