search for: lardconcept

...> You can do many things, document each node, and save xml with each > extension. > We?ve made it open source on Astricon 2015 you can extend it the way you > want. > > Hope it helps you. > > Best regards > > > > > On Mar 18, 2017, 12:50 +0100, Jonathan H <lardconcepts at gmail.com>, wrote: > > How are we all documenting complex dialplan? > > Is there something similar to Doxygen? > > I've got around 20 config files covering around 60 contexts and 40 > variables. Of course, I've maintained a basic list of the major stuff, > and...

So if someone has their own hardware and infrastructure but wants a software (not FreePBX but perhaps similar) what options do we have? Would like to virtualize it and not stuck with any one virtualization technology. Discuss... :) Travis -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ssions on that directory. Hmmm.... On 4 November 2016 at 21:50, John Kiniston <johnkiniston at gmail.com> wrote: > I'm able to use the FILE function to create files just fine. > > Set(FILE(${CALLFILE},,,al,u)=Extension: s) > > On Fri, Nov 4, 2016 at 2:26 PM, Jonathan H <lardconcepts at gmail.com> wrote: >> >> Seems I can write to an existing file, but is there really no way of >> creating a new file to log some data to, without reverting to AGI? >> (will be different for each caller ID) >> >> -- >> _________________________________...

...was started (after an agi_hangup/break etc was sent and the caller returned to dialplan) but it appears to persist for the duration of the call/channel. Hmmm.... On Mon, 23 Mar 2020 at 12:40, Joshua C. Colp <jcolp at sangoma.com> wrote: > On Mon, Mar 23, 2020 at 9:30 AM Jonathan H <lardconcepts at gmail.com> wrote: > >> Hope you're all well. >> >> I know we should be using https://community.asterisk.org/ but until >> someone lets Google know that it's moved, all the search results (and >> Asterisk's own search results) come from https://f...

...On Wed, 6 May 2020 at 07:21, Dan Jenkins <dan at nimblea.pe> wrote: > Hi Jonathan, > > I'd probably go down the external media route in the ARI now - you make > your own RTP server and provide your own RTP back to asterisk > > On Sun, 3 May 2020, 13:07 Jonathan H, <lardconcepts at gmail.com> wrote: > >> Way back in 2016 the only way to allow callers to listen in to a stream >> "at will" was to do the following: >> >> moh.conf >> >> [radio] >> mode=custom >> application=/usr/bin/mplayer https://example.com/st...

...t would indeed be wonderful, at this point I really do not care whether I need to use Python or Lua or JS. I was following http://zaf.github.io/asterisk-speech-recog/ but hit a road end with (for the lack of sane word ) copulating Google's Key On Wed, Jul 19, 2017 at 2:28 PM, Jonathan H <lardconcepts at gmail.com> wrote: > Yes! But I can only tell you if you can use Python, as I used Google's own > demo code. > > If you can hold on for half an hour, I'll remove personal info and put a > version up on Github if you're interested? > > > On 19 July 2017 at...

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. > > > &...

...missing a trick here and there's no better way to do those to things, is there any way to force Asterisk to NOT "optimize" those channels? On 9 November 2016 at 00:09, Richard Mudgett <rmudgett at digium.com> wrote: > > > On Tue, Nov 8, 2016 at 5:19 PM, Jonathan H <lardconcepts at gmail.com> wrote: >> >> Asterisk 14.1 >> >> Here's a bit of test dialplan, which works as expected and simulates >> exactly what I'm doing at the top of my large dialplan... >> >> [dial-pre-test] >> exten => s,1,NoOp() >> s...

...um.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent: Tuesday, April 18, 2017 09:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16 Feel free to take a look at https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/m aster/Asterisk-14-on-Ubuntu.md Ignore the bit about Voipfone and just skip to the "Install Asterisk" bit. I've used this same script with Asterisk 12,13 and 14 on Ubuntu 15,16 and 17 so this should work! Let me know how you get on. On...

...for # and it is quite easy to add all > other keys 1-9 and etc.... > > Then change code accordingly so script returns value of key. > > As far as I remember it wasn't hard. > > With kind regards, > > > Jurijs > > On Wed, Dec 6, 2017 at 4:50 PM, Jonathan H <lardconcepts at gmail.com> wrote: >> >> Thanks Jurijs, >> >> Yes, in fact I'm already using that, and it works fine. The problem >> here is that I cannot find a way of recording speech AND listening for >> a DTMF digit being pressed as an alternative. >> >&gt...

...about 500ms (10x more) Anyway, not to worry, a few seconds is fine, just thought I'd point out the discrepancy! Thanks again for the project/service Jonathan On Fri, 15 May 2020 at 15:52, Joshua C. Colp <jcolp at sangoma.com> wrote: > On Fri, May 15, 2020 at 9:39 AM Jonathan H <lardconcepts at gmail.com> wrote: > >> Hello! >> >> https://forums.asterisk.org/ is doing it again - "Content Encoding >> Error. An error occurred during a connection to forums.asterisk.org. >> Please contact the website owners to inform them of this problem". >...

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

...attr_opus(M) Support Level: external" However, while I managed to apt install xmlstarlet, I already had bash, and I can't find res_format_attr_opus. Any ideas? Many thanks, Jonathan > Regards. > > -- > Ludovic Gasc (GMLudo) > > 2018-01-27 2:06 GMT+01:00 Jonathan H <lardconcepts at gmail.com>: >> >> 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) Scro...

Hope you're all well. I know we should be using https://community.asterisk.org/ but until someone lets Google know that it's moved, all the search results (and Asterisk's own search results) come from https://forums.asterisk.org/ In most browsers, it's not displaying; in Firefox, it says: "Content Encoding Error The page you are trying to view cannot be shown because it

Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit being pressed as an alternative. That's where the problem lies. J.

How are we all documenting complex dialplan? Is there something similar to Doxygen? I've got around 20 config files covering around 60 contexts and 40 variables. Of course, I've maintained a basic list of the major stuff, and documented the code throughout, but it's grown to the stage where it needs to be better documented, have a proper flowchart etc. Talking of flowcharts, I see

I've not needed to do a dialplan reload for a while, so I don't know exactly which version is stopped working, but on 15.5, I'm not seeing ANY debug info at any debug level. So I'm not really sure how to find mistakes in the dialplan. This is all I get... how do I enable this debug mode to see the previous behaviour? Thanks asterisk -rvvvvvddddd (enters console) dialplan reload

Way back in 2016 the only way to allow callers to listen in to a stream "at will" was to do the following: moh.conf [radio] mode=custom application=/usr/bin/mplayer https://example.com/stream.mp3 -quiet -ao pcm:file=/dev/stdout -af volume=5,resample=8000,channels=1,format=alaw extensions.conf exten => radio,1,Verbose(1, Entered radio context) same => n,Set(VOLUME(TX)=1)

...her I need to use Python or Lua or JS. >> >> I was following http://zaf.github.io/asterisk-speech-recog/ >> but hit a road end with (for the lack of sane word ) copulating Google's >> Key >> >> >> >> On Wed, Jul 19, 2017 at 2:28 PM, Jonathan H <lardconcepts at gmail.com> >> wrote: >> >>> Yes! But I can only tell you if you can use Python, as I used Google's >>> own demo code. >>> >>> If you can hold on for half an hour, I'll remove personal info and put a >>> version up on Github if...

...l be easy. You'd use agi or something to navigate the object you create > and tts to describe current position. The hard part will be parsing the HTML > even though most HTML is broken :-) > > Kind regards, > > Matt Riddell > > On Oct 17, 2016, at 9:00 AM, Jonathan H <lardconcepts at gmail.com> wrote: > > Has anyone attempted making the web phone accessible? I can only find one > company which operated between 1996 and 2000. > > I was thinking, install Chrome with Chromevox, headless, on a server, and > use something like an AGI to send basic keyboard c...