similar to: [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,

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 Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Tried with both softphone (Ekiga) and snom IP phone, contact header > contains the public IP. but from header contains the private IP. after > OPTIONS method sent by the server. client sends an Register with expires 0. > Ok, did setting from_domain work? > > Best

On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Thanks for the reply. > > Yes my PABX is on the cloud when I try to register to the server, the > server sends registration OK with public address but OPTION method > includes the private address of the server in from header not the public > address. I have include

Hello, I carefully read [1] which details how backtrace files can be produced. Maybe this seems natural to some, but how can I go one step futher, and check that produced XXX-thread1.txt, XXX-brief.txt, ... files are OK ? In other words, where can I find an example on how to use one of those files and check by myself, that if a system ever fails, I won't have to wait for another failure to

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com> wrote: > > > On Wednesday, 14 September 2016, Madushan Geethanga < > mgliyanage.rc at gmail.com> wrote: > >> Hi, >> >> What is the equal option for externip in asterisk 13 with pjsip. I have >> tried >> >> external_media_address=XX.XX.XX.XX >>

@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

On Mon, Apr 25, 2016 at 11:11 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > 2016-04-25 18:14 GMT+02:00 George Joseph <gjoseph at digium.com>: > >> >> >> On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> >> wrote: >> >>> >>> >>> On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07

On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I've just discovered PJSIP 's support of set_var setting in pjsip.conf. >> Is this setting also supported in pjsip_wizard.conf ? >> On a fresh 13.8.2, it

Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get

Hello, >From Astricon 2019 notes [1], you can read "[a]ll 3 app_voicemail variants can now be built". What does it mean ? Is this change tied with a specific Asterisk version ? Is possible to change from ODBC to IMAP without re-compilation ? Is it also possible to mix mailbox types on a single system ? [1] https://wiki.asterisk.org/wiki/display/AST/Agenda+2019 Best regards

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

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

On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> > wrote: > >> Has anyone successfully used Mysql realtime PJSIP with Asterisk >> 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the >> following error now: >>

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

On Mon, Sep 12, 2016 at 3:01 PM, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 9/12/16 3:39 PM, George Joseph wrote: > > > > On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> >> wrote: >> >>> Has

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, Watching Snom 870's video (http://www.youtube.com/watch?v=9e8hPxX0oDU), you can see a new sidecar (phone extension) which seem very interesting. Has someone details on this extension ? Any release date or/and data sheet ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've just discovered PJSIP 's support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ? On a fresh 13.8.2, it doesn't seem but I may have missed somthing. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: