search for: gmludo

..."LG" <2540> secret = XXXXXXXXXXXXXXXXXXXXXXXXXX type = friend subscribemwi = no mohsuggest = default qualify = yes fromdomain=lg.allocloud.com fromuser=2540 If I use only [2540] as section name, I'll have a clash on the same Asterisk. Thanks for your answers. -- Ludovic Gasc (GMLudo) http://www.gmludo.eu/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150924/e83198c0/attachment.html>

Hi. thanks a lot for your reply. i will download the newer libical software. Could you elaborate on icalendar with google calendar config and calendar.conf, please? On Thu, Mar 15, 2018 at 3:00 PM, Ludovic Gasc <gmludo at gmail.com> wrote: > I never use caldav mode, always icalendar with Google Calendar. > > BTW, you use old versions of libical, Asterisk and Debian, I recommend you > to upgrade or install a new server with Debian Stretch: You will have > Asterisk 13, libical2 and security upgra...

On 10-08-16 08:52, Ludovic Gasc wrote: > > For WebRTC, I recommend you to use Asterisk 13+. > > Have a nice day. > > Ludovic Gasc (GMLudo) > http://www.gmludo.eu/ > > > Hello then why is there an option in sip.conf and rtp.conf " icesupport=yes" ?? This is no answer to my question. So again : what am I missing to get ICE support on my Asterisk 11.23.0 ?? Kind regards. -------------- next part ----------...

...hen a reload to the system and try to see: PBX> calendar show calendars Calendar Type Status -------- ---- ------ there is nothing shown over here. Off course i checked if my calendar is public but i have not an idea why it is not working. On Tue, Mar 13, 2018 at 10:23 PM, Ludovic Gasc <gmludo at gmail.com> wrote: > Hi, > > I recommend you to use calendar module of Asterisk: > https://wiki.asterisk.org/wiki/display/AST/Asterisk+Calendaring > > We are using in production since two years with several hundred calendars, > it works pretty well. > However, I strong...

Hi! Unfortunately pjsip is broken in Ubuntu Asterisk installed from repo. Yes I also thought to try with pjsip, just to know if it's also affected. I'll try to make a test next days. On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc <gmludo at gmail.com> wrote: > Hi Kseniya, > > You might test with chan_pjsip: We have less production experience with > chan_pjsip than chan_sip, however, for now, we are more and more confident > in this new stack while we're digging in documentation and we're testing on > p...

hi, is there somebody using systemd start script on fedora/centos7 + asterisk 13 in production? i have strange problem with high cpu usage when asterisk is started via systemd thanks for feedback p.s. systemd script is not in vanilla asterisk. only in fedora package info https://reviewboard.asterisk.org/r/2730/ -- --------------------------------------- Marek Cervenka

Ludovic Gasc wrote: > Hello, > > same sip username with realms is possible with Asterisk ? > I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and > now, Asterisk crashes. Did PJSIP crash in general (it's usually a build problem if that happens) or was it when you were experimenting with different realms and such? -- Joshua Colp Digium, Inc. | Senior

...it seems to change only the digest auth value during registration. I'm pretty new with PJSIP, however, I have the impression that you can't do that with Asterisk, each username in PJSIP must be unique accross an Asterisk instance. Is it correct ? Thanks for your answers. -- Ludovic Gasc (GMLudo) http://www.gmludo.eu/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150626/408da71e/attachment.html>

On 27 January 2018 at 09:27, Ludovic Gasc <gmludo at gmail.com> wrote: > Hi Jonathan, > > If you put the cursor on the line XXX, you will see what are the > dependencies are missing to enable the option. > In this case, it's certainly curl that is missing on your system. Ah, OK! No, it wasn't curl that was missing, but I...

Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined

​I need to launch a remote process at the machine that has the dialer. I could hard-code the IP address in a global variable, but It would be much more elegant if the dialplan would have a "manager" object where I could read "manager-->connected". ​ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. in my home office i operate my asterisk and have an IVR that has the business hours 9-5 and everytime i edit it to load the bank holidays (New Years eve, christmas, easter, whatever else). I would like to be able to load in the Asterisk's DB or in a file for all the year or years the planned holidays. Then it will be read from that file to operate accordingly. Is there a hint on how to run

...dard+Channel+Variables > > Nevertheless, the variable seems to be set in the Asterisk source code: > https://github.com/asterisk/asterisk/blob/13.10/main/bridge.c#L1222 > I see no issues open about that, do I need to open an issue ? > > Have a nice week. > -- > Ludovic Gasc (GMLudo) > http://www.gmludo.eu/ > > 2016-09-17 11:47 GMT+02:00 Jonas Kellens <jonas.kellens at telenet.be > <mailto:jonas.kellens at telenet.be>>: > > Hello > > a call goes out and is answered : > > [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dia...

Ok, thank you for the assistance! ??, 13 ???. 2017 ?. ? 16:38, Joshua Colp <jcolp at digium.com>: > On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote: > > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic > > and > > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same behavior. > > Joshua, maybe you can advice what can

In a fork of seanbright's opus patch for 13 there are further patches for Forward Error Correction and Package Loss Concealment, both of which ought to very useful in voip: https://github.com/traud/asterisk-opus Anybody used these patches ? Puzzled why they weren't committed to the main patch. sean

Before I got an log a ticket, can I just check I'm not doing anything wrong? In 15.2, to install Opus: 1) run `make menuselect` 2) Highlight "Codec Translators" and press enter. 3) Scroll down to "codec_opus" in the section labeled "External" 4) Press enter to select the codec if it is not already selected. ... at this point, I see XXX codec_opus and a

...tthew Fredrickson > > On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote: >> On 10-08-16 08:52, Ludovic Gasc wrote: >> >> For WebRTC, I recommend you to use Asterisk 13+. >> >> Have a nice day. >> >> Ludovic Gasc (GMLudo) >> http://www.gmludo.eu/ >> >> >> >> >> Hello >> >> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ?? >> >> This is no answer to my question. >> >> So again : what am I missing to get ICE supp...

Hi All We have about 15 different asterisk boxes around the place and on my list has been automate deployment updates and keep a revision history. They are mostly not publicly accessible, and external SIP access is closely firewalled , so updates happen straight away when its something like heartbleed, but take a while to trust/test new releases. Our boxes are Ubuntu LTS - mostly 14.04 at

Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

It means, AMI application is no more running or crashed or lost network connection with asterisk server. In such cases call is neither answered nor disconnected by Asterisk. I want to detect such state and jump to next dial plan to answer or reject the calls Regards Amit Patkar On September 20, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote: >On Sat, Sep 17,