similar to: Debug compile fails

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.

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

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 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 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

Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: root at Sangoma-Testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 What should i do? Thanks! -------------- next

Hi everybody, sometimes we have an Asterisk-crash, but no clue why this is happening, so I'm trying to make a coredump to analyse it. I compiled Asterisk 1.4.20.1 on CentOS 5.4 i386 with "DEBUG_THREADS" and "DONT_OPTIMIZE", then I start it with: # /bin/bash /usr/sbin/safe_asterisk This should do an "ulimit -c unlimited", but I entered it in the terminal again.

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

On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> wrote: > 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....):

Hi, I can cross compile asterisk 1.4.21 on arm (imx27) using ltib I want to cross compile the new version 1.6.2.13 but there is an error when I execute the commands : ./configure --build=i686-pc-linux-gnu --host=arm make menuselect The configure seems ok, I have the result info : *configure: Package configured for: configure: OS type : none configure: Host CPU : arm configure:

Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? 2 channels below are somehow associated, but how can I be 100% sure they are related in order to implement a redirect of the incoming call to another phone ("attended" call pickup respecting call/pickupgroups). Uniqueid seems to be a

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

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. > > > >

I was wondering why *foo at default should match '_*[0-9a-zA-Z].*0.' in 1.6.13. Who is making the parse error, * or me? CLI> dialplan show *foo at default '_*[0-9a-zA-Z].*0.' => 1. NoOp(${EXTEN}) [pbx_config] 2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config] 3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]

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

Hi Kevin! Thanks very much for your reply! Much appreciated! So I just have a remaining question from this, if the with-ssl is not mandatory to have the encryption support, what is it actually used for? Maybe it is some old flag which is not needed anymore and so can be ignored for now and possibly removed from the configure/makefile stuff for future releases? Kind regards, Patrick Wakano On

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

> On Wed, May 31, 2017 at 12:36:47PM -0700, Steve Edwards wrote: >> I want to capture all SIP messages. >> >> I have about 30 hosts in about 6 colos. >> >> My first thought was dumpcap, but the output file name format bugs me. >> >> What do you use for long term SIP capture? On Wed, 31 May 2017, Daniel Tryba wrote: > What bugs you about the output

On Wed, 31 May 2017, Daniel Tryba wrote: > On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote: >>> What bugs you about the output format? >> >> It's been a while, but as I recollect, it included the date/timestamp in the >> file name of the 'ring buffer' which meant that each time the host was >> rebooted, dumpcap didn't know the

Anyone ever noticed that the reported holdtime is wrong when there are different priorities? Also talktime is 0, but for the moment I don't care. "queue show test" reports: test has 23 calls (max unlimited) in 'ringall' strategy (193s holdtime, 0s talktime) [...] Callers: 1. Local/351 at default-8828;2 (wait: 3:32, prio: 15) 2. Local/351 at default-8361;2