asterisk users - Oct 2014 - Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes

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

-Dave

On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton <
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, 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
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