search for: nimblea

Asterisk 18 will have support based on this asterisk update Matt F did for CommCon's sponsor slots https://youtu.be/eas1csaX-wc On Sun, 12 Jul 2020, 22:44 Steve Edwards, <asterisk.org at sedwards.com> wrote: > On Sun, 12 Jul 2020, Saint Michael wrote: > > > WORLDWIDE EMERGENCY > > Again? > > > The code below needs to be executed before any SIP or PJSIP call

Hello, I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only problem until now which remained was that if dtls_rekey was set to the value other than 0, call hanged up when using chrome after the time where dtls_rekey was set. I suppose that "bad media description" shown in Chrome's window which causes call to fail, has appeared with Chromes newer versions

On 13.07.20 at 00:17 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins <dan at nimblea.pe> wrote: > >> Asterisk 18 will have support based on this asterisk update Matt F did for >> CommCon's sponsor slots >> >> https://youtu.be/eas1csaX-wc >> >> > As well support will go into Asterisk 16 and 17 as well. It's just been > under a...

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 asterisk > > On Sun, 3 May 2020, 13:07 Jonathan H, <lardconcepts at gmail.com> wrote: > >> Way back i...

On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins <dan at nimblea.pe> wrote: > Asterisk 18 will have support based on this asterisk update Matt F did for > CommCon's sponsor slots > > https://youtu.be/eas1csaX-wc > > As well support will go into Asterisk 16 and 17 as well. It's just been under active development so we've been wai...

On Sun, Jul 12, 2020 at 11:37 PM Michael Maier <m1278468 at mailbox.org> wrote: > On 13.07.20 at 00:17 Joshua C. Colp wrote: > > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins <dan at nimblea.pe> wrote: > > > >> Asterisk 18 will have support based on this asterisk update Matt F did > for > >> CommCon's sponsor slots > >> > >> https://youtu.be/eas1csaX-wc > >> > >> > > As well support will go into Asterisk 16 and...

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)

First things first, upgrade from 13 - WebRTC has moved a long a lot since then. If you can't upgrade everything to 13 then run another asterisk specifically for WebRTC and bridge to your other Asterisk Is this just an audio conference? On Sun, Apr 26, 2020 at 10:21 PM Teijo <g.aloitus at gmail.com> wrote: > Hello, > > > Has somebody get combination Asterisk (I'm

I honestly couldn't tell you if it would resolve it but there aren't many people going to be willing to help problem solve anything if you're running 13 - you'll get more support on 17 for example. Very easy to bring up a new instance or VM in the grand scheme of things to test the theory and get it working on most recent version of Asterisk On Tue, Apr 28, 2020 at 11:37 AM

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]