similar to: How to associate a custom script to "core stop now"

2016-09-02 20:40 GMT+02:00 George Joseph <gjoseph at digium.com>: > > > On Fri, Sep 2, 2016 at 9:34 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I had a recent case where Asterisk stopped due to a segfault. >> This reminded me that being sure that whenever such issue occurs, it's >> useful to have a core file or various

Hi, I'm using asterisk 1.6.1 and AEL2. I'm trying to find the best way to write my own custom functions ? At the moment, I'm using this pattern (extensions.ael) : context foo { 123 => { &myfunc(123456); NoOp(${GOSUB_RETVAL}); }; macro myfunc (arg) { Return (${arg}); } 1. First, I keep getting warnings like Warning: file /etc/asterisk/extensions.ael, line

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,

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

2016-09-06 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. >>

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, 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

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Hi, I'm trying to install spandsp from source in a Debian Lenny system. I did : cd /usr/src wget http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre5.tgz tar xvf spandsp-0.0.6pre5.tgz cd spandsp-0.0.6 ./configure make make install When doing this, spandsp warns me that librairies are installed in /usr/local/lib with this message : "Libraries have been installed in:

Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090627/37b93684/attachment.htm

Hi, I feel I've read a thread about this previously but I couldn't find it. Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. So that, you could

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

This happens mysteriously & randomly. If asterisk was killed and restarted, it often gives this error myast*CLI> core stop now No such command 'core stop now' (type 'core show help core' for other possible commands) Any hint Thanks Jim

Hi, I need to replace digital handsets in offices where there cabling is appareantly not Ethernet-compliant. Today's usage is to press a key to toggle between private ou public line before issuing an outgoing call. Are you aware of a DECT handset (to overcome cabling limitations) that mimic this line-key behaviour ? For instance, acceptable behaviours would be to dial number string and press

2009/3/17 Olivier <oza-4h07 at myamail.com> > > > 2009/3/16 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I'm rather new to this domain so I may be doing stupid things without >> being concious of that. >> >> I've got a Patton MATA I'm trying to setup as T.38 fax adapter. >> Whenever I connect a fax machine (Dell