similar to: Configuration management and update deployment - what do you use?

Hello, In older version of freebpx if you write somethng manually in the configuration files it was flushed by amp, i.e. you can configure it through the interface only. Is this this thing still present in freepbx? _________________________________________________________________ Don't just search. Find. Check out the new MSN Search!

Hi, How have the same sip username in several realms ? For now, I must add the realm prefix in the sip username of chan_sip. For example: [lg_2540] amaflags = default call-limit = 10 host = dynamic language = en_US context = lg_default callerid = "LG" <2540> secret = XXXXXXXXXXXXXXXXXXXXXXXXXX type = friend subscribemwi = no mohsuggest = default qualify = yes

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

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

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 ??

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

Hi. Thanks for the idea for calendar, it sounds better. i did not manage to make it work though. i am running debian 8 32 bit with asterisk 11.25.3. I have installed the packages libneon27-dev & libical-dev then in /etc/asterisk the file calendar.conf has the following entries: [Gcalendar] type=caldav url=https://www.google.com/calendar/dav/atuxnull at gmail.com/events/ user=atuxnull at

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

Hi, In PJSIP configuration, I thought that "from_domain" parameter in a endpoint permits to have two SIP peers with the same usernames with a different domain. I've tested at the transport level, I see no changes. I've also tested with realm parameter in auth configuration, it seems to change only the digest auth value during registration. I'm pretty new with PJSIP,

​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

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

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 think the way it was phrased confused me. It

I had to restart one of my routers tonight and since then shorewall on it has been dropping SIP packets coming in from one machine instead of forwarding them to the freebpx server. Shorewall:net2all:DROP:IN=eth0 OUT= MAC=<removed> SRC=<my home network external ip> DST=<server network external ip> LEN=575 TOS=0x00 PREC=0x20 TTL=78 ID=230 PROTO=UDP SPT=5061 DPT=5060

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

I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

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