Displaying 20 results from an estimated 300 matches similar to: "CPU usage vs compiler flags"
2016 Sep 07
2
[SOLVED] Re: Feature Request: what about "core stop panic" ?
2016-09-06 17:48 GMT+02:00 Tzafrir Cohen <tzafrir.cohen at xorcom.com>:
> On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote:
> > On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote:
>
> > > Where should core file be created when Asterisk is run as a daemon by
> > > asterisk user and group ?
> > > Is there a setting I
2016 Sep 08
2
[SOLVED] Re: Feature Request: what about "core stop panic" ?
I think were getting closer:
I did:
- I edited /etc/default/asterisk to include :
AST_USER="root"
AST_GROUP="root"
# systemctl daemon-reload
# systemctl start asterisk
# ps aux | grep asterisk
root 3602 7.1 2.5 60332 26012 ? Ssl 16:00 0:03
/usr/sbin/asterisk -U root -G root -g
# rasterisk
# pkill -SEGV asterisk
Then console showed:
Segmentation error (core
2016 Sep 06
5
[SOLVED] Re: Feature Request: what about "core stop panic" ?
On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote:
> Hello,
>
> After testing "pkill -SEGV -f /usr/sbin/asterisk" on Debian Jessie
> platform, I've got several questions :
>
>
> 1. When I issue a "cd /tmp; asterisk -cvvvvvvvvvvvg -U asterisk -G
> asterisk" command, and then issue a "pkill -SEGV asterisk" command,
2010 Sep 24
2
Debug compile fails
Somehow I can't get 1.6.2.13 to compile with DEBUG_CHANNEL_LOCKS.
Downloaded latest tgz and extracted
$ ./configure
$ make menuselect
(select the needed options from compiler flags)
$ grep DEBUG_CHANNEL_LOCKS menuselect.makeopts
MENUSELECT_CFLAGS=DONT_OPTIMIZE LOADABLE_MODULES DEBUG_CHANNEL_LOCKS
MALLOC_DEBUG
$ make && make install
$ asterisk && asterisk -rx "core show
2010 Sep 14
6
Spontaneous reboots on asterisk 1.6.2.11
Hello list,
has anyone else also noticed spontaneous reboots ?!
I noticed this today and also yesterday. Can't really see if there is a
fixed time between the reboots.
Normally al of my SIP peers are registered. When I put up the CLI today
I saw that a lot of SIP accounts where UNREACHABLE and needed to
register again (what they slowly did).
These are realtime SIP peers that reside on
2008 Jun 03
3
Asterisk 1.4.20.1 with bad gsm file playback
Hi All,
I'm stumped on this and I looking for some clues to fix this.
This is a new install of Slackware 12.1 onto an IBM x330 Server.
Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just
fine, but when I play the gsm files the audio quite choppy. And, the files
produced from the MixMonitor don't even record any audio other than noise.
I have a hard drive from
2018 Jun 26
2
Asterisk crashing on AAAA lookup
I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so often
asterisk crashes and then restarts. I am not seeing any core dumps on the
box. The only I thing I see every time is a second before Asterisk crashes
there is a AAAA lookup for the boxes hostname. As soon as it gets the
response I see that asterisk is restarting. Any idea what would cause this
and how would get a dump or
2011 Nov 23
1
DONT_OPTIMISE, BETTER_BACKTRACES and performance
Hi
How much impact on performance do DONT_OPTIMISE and BETTER_BACKTRACES
have on a busy (13000+ entries in cdr for yesterday) server?
I'm trying to decide whether to have them on in case of crashes or not.
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2009 Mar 29
1
DUNDi broken in asterisk 1.4-svn?
Hi Guys,
since about two weeks pbx_dundi.so from svn segfaults when i load it, 1.4.24 release works fine on the same box. Can someone tell me if that's something weird with my Fedora8 system or a possible bug in svn?
Program terminated with signal 11, Segmentation fault.
#0 0x00000000 in ?? ()
(gdb) bt
#0 0x00000000 in ?? ()
#1 0x0881d00c in dundi_encrypt (trans=0x985e6e0, pack=0x985b5e0)
2016 Nov 29
5
Any reason Asterisk won't start without a rebuild on a cloned VPS?
On Tue, Nov 29, 2016, at 07:15 AM, Barry Flanagan wrote:
> On 29 November 2016 at 10:56, Jonathan H <lardconcepts at gmail.com> wrote:
>
> > Any ideas why a VPS, cloned from another instance (DigitalOcean
> > "droplets" if it matters), won't run on the new instance?
> >
> > Everything else is the same; region, memory, disk, hypervisor version etc.
2016 Nov 29
2
Any reason Asterisk won't start without a rebuild on a cloned VPS?
Any ideas why a VPS, cloned from another instance (DigitalOcean
"droplets" if it matters), won't run on the new instance?
Everything else is the same; region, memory, disk, hypervisor version etc.
And everything else runs, just not Asterisk, unless I do a make
distclean in the /usr/src/asterisk directory, rebuild, and then it
runs just fine.
I'd understand if I was moving it
2018 Feb 21
2
Asterisk crash on core show channel
Hello Asterisk list,
I am facing some Asterisk crashes which are consistently pointing to the
same backtrace, which is the following (using DONT_OPTIMIZE,
BETTER_BACKTRACES and MALLOC_DEBUG):
Thread 1 (Thread 0x7f1f08be8700 (LWP 1767)):
#0 0x00007f1f9bed3395 in __strcasecmp_l_sse42 () from /lib64/libc.so.6
#1 0x00000000004a91ca in cdr_object_get_by_name_cb ()
#2 0x0000000000463c60 in
2007 May 15
0
PATH_MAX' undeclared here (not in a function) in asterisk!
hello, James FitzGibbon:
thank you for your help. i am very new to arm-linux and embedded linux. i think what you said is right. i am not very sure the steps i taken are correct. i post it here and please give me some help. it might be help other arm-linux users too. i installed all necessary libraries in my linux. if i just install asterisk under my linux. there is no problem. but when i
2010 Mar 25
1
Static linking
Hi,
we have a problem with Asterisk that is described in https://issues.asterisk.org/view.php?id=15915. According to the last post in the bug report, a workaround is using of static linking. When I tried (I
enabled option Compiler Flags/STATIC_BUILD) it I got the following error message:
/usr/bin/ld: /usr/lib/gcc/x86_64-linux-gnu/4.1.2/crtbeginT.o: relocation R_X86_64_32 against
2015 Apr 29
2
Asterisk 1.8.32.3 chan_sip deadlock
Hello asterisk-users,
We've been having intermittent issues with chan_sip - it stops responding
to cli requests, trying to reload chan_sip from cli doesn't seem to have
any effect, initiated calls carry on for a short period, but no new SIP
requests are processed ('sip show channels' hangs forever, server stops
responding to SIP OPTIONS, or any other SIP messages). We have updated
2017 Feb 14
2
Advices when Asterisk segfaults and nothing useful in logs
Hello,
I've got a 13.13.1 system using PJSIP stack on debian Jessie.
It runs from 50 to 100 simultaneous calls (so 100 to 200 PJSIP channels)
all day long.
>From time to time, roughly meaning once a month, it segfaults with lines
(from dmesg -T output) like this:
asterisk[1160]: segfault at 7efffffe ip 00000000005881d6 sp
00007fec95c33910 error 4 in asterisk[400000+2a2000]
Debug level
2018 Jun 27
2
Asterisk crashing on AAAA lookup
On Tue, Jun 26, 2018 at 7:59 PM, Richard Mudgett <rmudgett at digium.com>
wrote:
>
>
> On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender <dovid at telecurve.com> wrote:
>
>> I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so
>> often asterisk crashes and then restarts. I am not seeing any core dumps on
>> the box. The only I thing I see
2018 Feb 21
2
Asterisk crash on core show channel
Thanks for you answer Marcus,
So maybe this means some bug was fixed? Anyone aware of something related?
>From the release notes, I couldn't find any direct change that could fix
this....
Thanks,
Kind regards,
Patrick Wakano
On 21 February 2018 at 20:29, Marcus Kvarsell <Marcus.Kvarsell at fogwise.se>
wrote:
> Hello, i found upgrading to asterisk 15 helped.
>
>
>
>
2014 May 24
1
"transmit_silence" not properly recognized on 1.8 ?
Hello,
I've got a problem at the moment, that setting "transmit_silence = yes"
seems to have no effect on Asterisk 1.8-Certified.
Although it's enabled and "core show settings" confirms, that it is
really enabled, there are no RTP packets sent by Asterisk when waiting
for DMTF input or when "Wait()" is called.
Also, there seems to be a small gap of 2 or 3
2014 Aug 14
1
Possible handle leak in PJSIP
I have been seeing errors saying the Asterisk cannot establish an RTP
connection, so I did this:
lsof -i -n -P | grep asterisk | wc -l
10483
but I have only
Asterisk 11 has 1 open calls
Asterisk 12 has 21 open calls
Asterisk 14 has 19 open calls
Asterisk 15 has 22 open calls
Asterisk 16 has 15 open calls
Asterisk 17 has 15 open calls
Asterisk 30 has 71 open calls
Total
164 active calls
The