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

Asterisk Team (JIRA) noreply at issues.asterisk.org
Fri Sep 4 12:01:33 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25337?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227477#comment-227477 ] 

Asterisk Team commented on ASTERISK-25337:
------------------------------------------

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> 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
>            Assignee: 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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