similar to: Old Asterisk forums not working

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". Which is odd, as the Qualys test seems to pass, only losing a point for supporting TLS 1.0. But I know it's not just me because Pingdom can't read the page, either.

Thanks Joshua; Working now - but regarding the slow wiki, it's normally 5-10 second page load for me in the UK, but it was closer to 20 seconds earlier. https://tools.pingdom.com/#5c862c04ad800000 Seems to be back to around 5 seconds now, but I notice that if I run a Pingdom page test from a US server, the page loads in about 1.3 seconds. I wonder if there might be a misconfigured edge

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://forums.asterisk.org/ > > In most browsers, it's not

Hi, When I go to my homepage all is well, however when the Pingdom bot arrives (every few seconds!), it throws the following error: ActionController::RenderError occurred in home#index: You called render with invalid options : available [RAILS_ROOT]/vendor/rails/actionpack/lib/action_controller/base.rb:808:in `render_with_no_layout'' My index action doesn''t call render, it

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". > > Which is odd, as the Qualys test seems to pass, only losing

Does anyone know of an online tool or program that would call our phone numbers to confirm that they are up? What I'm imagining is something like pingdom.com (a ping checker for web services), except for the phone system. It would call us, confirm an answer, then hang up. If no answer, it would send an e-mail to our cell phones so we knew about it right away. It seems like there would be a

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:

Hi all, I'm setting HTTP monitoring app, to check the availability of my stream, like pingdom or uptimerobot. After test this, I see that always see 200 ok inclusive when my stream is not online, and of course this tools don't send the alerts. some reason to do this? curl -X GET -I http://icecast/stream HTTP/1.0 404 Not Available Content-Type: text/html curl -X HEAD -I

Hi, thanks - that looks really good! I was about to embark on some non-visual stuff using Ragic, but this looks great. Is there a binary anywhere, or any instructions to compile? I've never compiled C# code before, and although a quick google suggests it shouldn't be too hard, I might need to know a few things like what version of .net it should be compiled with. The readme just points

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

That's just what I'm using, John. But I'm getting (eg) [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:449 file2format: Cannot open '/home/logs/anonymous.txt': No such file or directory [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:949 file_write: File '/home/logs/anonymous.txt' not in line format Asterisk is running as root (yeah, I know!), and has

Hi Jonathan Thanks ! That 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

Thanks Dan - might have to scratch my head over that one for a while! The phrase "you make your own RTP server" has made me all twitchy ;) Jonathan 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

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.

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

Hey; Thank you very much. I was able to install asterisk from your link. One other question. How are you starting asterisk? Do you use an init script or systemd? Do you think that you could share the script you use? Thanks Again; John V. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent:

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

Thank you - that makes sense. I've seen something about swapping and optimizing channels on the console, but I didn't realise "optimize" meant "not do what you wanted". OK, so here's why I'm dialling anything at all: The first dial is because I MUST limit the incoming call to less than 60 minutes. The second dial, which carries the gH option, is because I