similar to: Asterisk 1.8.32.3 chan_sip deadlock

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers I was getting this : [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf after fix global issue

Ok , digging more into this i could see that (timers=no) and (timers=forced) not work asterisk not pay attention to this options when is reloaded cli not say anything and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint. I wonder to force asterisk to refresh the session in some cases when is needed . pjsip is able to

I'd like to be able to save the choices made in menuselect in a way that they can be tracked in a CM system and applied to a later release of Asterisk using an automated tool like Ansible. What's the best way to do that?

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, 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.

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

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

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

Howdy, I built an LXC container with an "image" of asterisk 11.18 precompiled and installed. It runs fine on the dev platform, which is a Dell R320 running Ubuntu 14.04LTS. I shutdown the container, tarred it up, and untarred on a Dell PE1850, also running Ubuntu 14.04LTS. The container itself is Ubuntu 14.04LTS. Both platforms as far as I know are amd64. The container boots

Hi, i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing "XXX" -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 am upgrading an asterisk server from 1.2.4 to 1.4. i've installed libpri 1.4 i've installed zaptel 1.4 I've installed the new version of misdn with the script of beronet. i use this configure script: ./configure --with-postgres=/usr/local/pgsql then: make menuselect [*] 1. cdr_csv [*] 2. cdr_custom [*] 3. cdr_manager XXX 4. cdr_odbc [*] 5. cdr_pgsql XXX 6. cdr_radius

HI All I have a need to create a VM - install asterisk from source - then move the VM to a physical box. When I do this - I get illegal instruction when starting asterisk. So I recompile on the physical box and then everything is fine. Problem is there are a number of these physical boxes. The VM is Intel box (host) and the physical box is a celeron. So something is not right there. What would

Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54]

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

As posted on blogs.digium.com today: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ the Asterisk project has changed providers for Music-On-Hold (MOH) content distributed with/for Asterisk. In addition to the change for future Asterisk releases, we have also opted to rebuild historical releases with the new MOH content, in an effort to eliminate unnecessary distribution of

As posted on blogs.digium.com today: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ the Asterisk project has changed providers for Music-On-Hold (MOH) content distributed with/for Asterisk. In addition to the change for future Asterisk releases, we have also opted to rebuild historical releases with the new MOH content, in an effort to eliminate unnecessary distribution of

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 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)

The 'Make sounds' routine into Makefile doesn't seem to "pre-fetch" all of the audio tarballs. Is this an oversight or is there a strategic reason for it? Specifically it doesn't seem to fetch the MOH tracks for selected codec's. For example, during the most recent 1.6.1 update, the g.722 (asterisk-moh-opsound-g722.tar.gz) wasn't downloaded until "make