[asterisk-bugs] [JIRA] (ASTERISK-25337) Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25337?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-25337:
------------------------------------

    Attachment: rusty_pjsip.txt
                rusty_extensions.txt

[~gtj] it took me a while to get back to this. I pulled the latest 13 from git and reproduced it again (Asterisk GIT-13-c344fb0.)

I attached rusty_pjsip.txt and rusty_extensions.txt with the config I used.

Register a phone to each pjsip endpoint and then have ALICE call BOB at 6002, Asterisk should crash every time.

> Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub
> ---------------------------------------------------------------------
>
>                 Key: ASTERISK-25337
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25337
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip, Resources/res_pjsip
>    Affects Versions: SVN, 13.3.0, 13.5.0
>         Environment: VM built on VMware vcenter, running on a SSD based SAN
>            Reporter: Jacques Peacock
>            Assignee: Rusty Newton
>         Attachments: backtrace.txt, full.txt, messages.txt, rusty_extensions.txt, rusty_pjsip.txt
>
>
> 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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