similar to: 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

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

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

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

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. > > > >

Hello list, Hope you are all doing well! I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and I wonder if someone can put some light on it. Log history short, install_prereq fails to install the packages (not sure how important they actually are....): speexdsp-devel, gmime-devel, uriparser-devel, iksemel-devel, uw-imap-devel, hoard Then, I am running the following commands

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

On Tue, Feb 14, 2017 at 2:51 PM, George Joseph <gjoseph at digium.com> wrote: > > > On Tue, Feb 14, 2017 at 10:21 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> 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. >>

On Tue, Jun 26, 2018 at 7:06 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 06:57 PM, Dovid Bender wrote: > >> Hi, >> >> My dialplan looks like this: >> [from-external] >> >> Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) >> Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) >> Exten => _X.,n,Noop(CALLED

On Fri, Feb 17, 2017 at 5:17 AM, Olivier <oza.4h07 at gmail.com> wrote: > Hi George, > > How does ast_coredumper compare to ast_grab_core ) ? > Is it worth learning to use both or shall favor one ? > > PS: As I don't know either program, yet, my question may seem silly. > Please, forgive me for this > Not silly at all. ast_grab_core actually kills asterisk to

Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent Hylafax server using softmodems: Noticed this in the Asterisk log when trying to send a fax from Hylafax to Asterisk: [Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp: Multiple audio streams are not supported I've googled a few asterisk tickets that may suggest that yes, multiple audio streams are not

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

On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > Doug, > > I tried that as well. Even with my dialplan looking like this: > > > > Ordering by includes works for me under Asterisk 11 and 13 > > What does the output of the below show? > > dialplan show from-external > >

I have been following the "Asterisk CLI unresponsive" thread and saw the suggestion to enable BETTER_BACKTRACES when I went in to do that it said it requires libbfd.. When I went to add that from yast I get no packages by that name. I am on openSUSE an ideas if this is part of another package or how I would get it? Thanks Bryant -------------- next part -------------- An HTML

Paddy, It's pretty easy to spot from the CLI. A voicemail gets called. And the screen basically stops scrolling from there. Eventually you'll get the "Task processors exceeded 500 queued tasks" or something like that. And maybe channels attempting to hangup due to lack of RTP (If you have no-rtp timers configured). Once you find the problem mailbox, You can call it via any

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

Hi, I noticed that libvirt-lxc will be deprecated for RedHat: "Future development on the Linux containers framework is now based on the docker command-line interface. libvirt-lxc tooling may be removed in a future release of Red Hat Enterprise Linux (including Red Hat Enterprise Linux 7) and should not be relied upon for developing custom container management applications." -

Hi all, I just upgraded a production server to asterisk 1.4.25, compiling with the following: [*] 1. DONT_OPTIMIZE [*] 2. DEBUG_CHANNEL_LOCKS [*] 3. DEBUG_THREADS [*] 4. DEBUG_FD_LEAKS [ ] 5. LOW_MEMORY [*] 6.

Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

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)