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


 
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dfullertasterisk at sh...
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PostPosted: Thu Oct 23, 2014 3:32 pm    Post subject: [asterisk-users] Ast 13 beta 3 - Segfault when calling on pj Reply with quote
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

--
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mjordan at digium.com
Guest
PostPosted: Thu Oct 23, 2014 4:00 pm    Post subject: [asterisk-users] Ast 13 beta 3 - Segfault when calling on pj Reply with quote
On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton <dfullertasterisk@shorelinecontainer.com (dfullertasterisk@shorelinecontainer.com)> wrote:
Quote:
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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dfullertasterisk at sh...
Guest
PostPosted: Fri Oct 24, 2014 8:47 am    Post subject: [asterisk-users] Ast 13 beta 3 - Segfault when calling on pj Reply with quote
On 10/23/2014 05:00 PM, Matthew Jordan wrote:
Quote:


On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton
<dfullertasterisk@shorelinecontainer.com
<mailto:dfullertasterisk@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


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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