[asterisk-users] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes

[Dave Fullerton]

On 10/23/2014 05:00 PM, Matthew Jordan wrote:
>
>
> On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton
> <dfullertasterisk at shorelinecontainer.com
> <mailto:dfullertasterisk at shorelinecontainer.com>> wrote:
>
>     Hello all,
>        I'm setting up a couple of test boxes and I'm running into a
>     problem. What I need help with is determining whether I'm going
>     something wrong or if I need to post a bug report. I have two
>     asterisk 13.0-beta 3 machines set up with extensions connected to
>     each as such:
>
>     3700 ----> AST-A  <------> AST-B <---- 3800 & 3801
>
>     When I place a call from 3800 to 3700 or the other way around ,
>     asterisk seg faults on both machines at roughly the same time. All
>     connections are done using PJSIP.  The crash occurs when the ringing
>     extension is answered.
>
>     If I set (directmedia=no) OR (directmedia=yes & t38_udptl=yes) on
>     the trunk then the call completes fine. All phones and servers are
>     on the same LAN with no firewalls active.
>
>     The trunk between AST-A and AST-B is configured like this in
>     pjsip.conf and is identical on both machines:
>
>     [transport-lan]
>     type=transport
>     protocol=udp
>     bind=0.0.0.0
>     tos=af31
>
>     [pbxbeta]
>     type=endpoint
>     disallow=all
>     allow=g722
>     allow=ulaw
>     transport=transport-lan
>     context=phone-level3
>     aors=pbxbeta
>     send_rpid=no
>     send_pai=yes
>     trust_id_inbound=yes
>     trust_id_outbound=yes
>     direct_media=yes
>     direct_media_glare_mitigation=__outgoing
>     ;direct_media_method=update
>     tos_audio=46
>     tos_video=34
>     t38_udptl=no
>     t38_udptl_nat=no
>
>     [pbxbeta]
>     type=aor
>     contact=sip:{remote IP address}:5060
>
>     [pbxbeta]
>     type=identify
>     endpoint=pbxbeta
>     match={remote IP address}
>
>
>     The phones have the following set in pjsip.conf (snippet):
>     type=endpoint
>     disallow=all
>     allow=g722
>     allow=ulaw
>     transport=transport-lan
>     send_rpid=no
>     send_pai=yes
>     direct_media=yes
>     tos_audio=46
>     tos_video=34
>
>     Is there something I'm doing wrong here?
>
>     Thanks
>
>
> Asterisk shouldn't crash.
>
> Please file a bug report ASAP at issues.asterisk.org
> <http://issues.asterisk.org>, with a properly generated backtrace:
>
> https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
>

Created: https://issues.asterisk.org/jira/browse/ASTERISK-24448

Let me know if you need any more information.

Thanks

-Dave