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

[Dave Fullerton]

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