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

Matthew Jordan mjordan at digium.com
Thu Oct 23 16:00:17 CDT 2014


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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