[asterisk-bugs] [JIRA] (ASTERISK-25337) Asterisk Crash on PJSIP Add P-Asserted-Identity

Richard Mudgett (JIRA) noreply at issues.asterisk.org
Fri Aug 21 10:58:32 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25337?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227329#comment-227329 ] 

Richard Mudgett commented on ASTERISK-25337:
--------------------------------------------

Thank you for your bug report. In order to move your issue forward, we require a backtrace[1] from the core file produced after the crash. Also, be sure you have DONT_OPTIMIZE enabled in menuselect within the Compiler Flags section, then:

make install

After enabling, reproduce the crash, and then execute the backtrace[1] instructions. When complete, attach that file to this issue report.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace



> Asterisk Crash on PJSIP Add P-Asserted-Identity
> -----------------------------------------------
>
>                 Key: ASTERISK-25337
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25337
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 13.3.0
>         Environment: VM built on VMware vcenter, running on a SSD based SAN
>            Reporter: Jacques Peacock
>
> We have an asterisk system configured to an external SIP trunk using the PJSIP driver using UDP as the transport. We use ael for our dialplan. 
> Asterisk is installed using the Digium repository, we do not compile it as we run approx 10 asterisk servers in various configurations, so we use the repos to make synchronising versions straightforward.
> Example endpoint configuration:
> {noformat}
> ;======ENDPOINT
> [testtrunk]
> type = endpoint
> context = ael-incoming-sm
> disallow = all
> allow = alaw
> transport=udptrans
> direct_media = yes
> direct_media_glare_mitigation = outgoing
> from_user = ourserver
> from_domain = ourdomain.local
> tos_audio = ef
> language = en
> aors = myaors
> send_pai = yes
> {noformat}
> Calls arrive from the remote trunk with the P-Asserted-Identity header populated.
> If send_pai is set to yes in the endpoint configuration, then attempting to add the header manually causes asterisk to crash with a segmentation fault:
> {noformat}
> Dial(PJSIP/111111 at testtrunk,,b(ael-setheaders^setheaders^1));
> context ael-setheaders
> {
> // Set SIP headers for the outgoing channel
> setheaders =>
> {
> Set(PJSIP_HEADER(add,P-Asserted-Identity)=sip:01234456789 at domain.local);
> Return();
> }
> }
> {noformat}
> If send_pai is set to no, then the command works as expected.
> I would not expect a crash to be the normal behaviour here, I would normally expect either a CLI error to occur or the set command to succeed



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