Re: [asterisk-users] Astricon 2012 presentations

Hi, As far as I'm aware the videos are still being produced and there's no definitive list anywhere for the slide decks. However, my one is here: http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614 Dan Jenkins -- Dan Jenkins - Senior Web

Re: [asterisk-users] How to MessageSend to a SIP from AMI Or CLI?

the unique current channel. The Asterisk wiki at wiki.asterisk.org is pretty good at showing you what's required, although their examples could be better, Hope this helps, Dan Jenkins On Nov 18, 2012, at 6:36, Face falaz...@gmail.com wrote: Hello all, I am running Asterisk 10.10.0 and I can

Re: [asterisk-users] Asterisk for Razberry Pi

On 2 January 2013 16:16, Chris Gentle gent...@gmail.com wrote: Does anyone know of any asterisk 11 packages for the Pi? I ended up compiling it myself this weekend. Took a while. Take a look at http://www.raspberry-asterisk.org/ :) --

Re: [asterisk-users] OPUS support in Asterisk 13

On Thu, Mar 24, 2016 at 2:50 PM, Tzafrir Cohen wrote: > On Thu, Mar 24, 2016 at 09:36:40AM -0500, Matt Fredrickson wrote: > > On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai > wrote: > > > Hi all, > > > > > > Sorry if this has been asked before. I

Re: [asterisk-users] http.conf - haproxy

On Mon, Aug 22, 2016 at 11:53 AM, Marek Červenka wrote: > hello, > > is it possible move asterisk http server behind haproxy (haproxy as SSL > endpoint, asterisk http only) > > any examples? > > my current http.conf > > [general] > enabled=yes > bindaddr=0.0.0.0 >

Re: [asterisk-users] Problem setting up ssl connection

On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens wrote: > Hello > > > I keep getting the following error when trying to connect to the Asterisk > server using AMI : > > $socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5); > > Erorr on CLI : > > [Oct 26

Re: [asterisk-users] Softphone with by-default AutoAnswer active?

On Thu, Oct 13, 2016 at 12:19 PM, Mandar Khire wrote: > Hi, > I know that This is not Asterisk related question but then also I ask this > question here due to Asterisk users know about softphones & here lots of > user present. > > Question:- > > I am looking for Softphone

Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

On Fri, Apr 7, 2017 at 9:44 PM, Teijo wrote: > 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

Re: [asterisk-users] Can someone provide some insight on WebRTC vs a generic SIP library in a browser?

Hi Dan, The SIP library they're talking about uses SIP over Websocket as a signalling protocol but uses WebRTC connections to Asterisk for media. Dan On Tue, Sep 11, 2018 at 5:12 PM, Dan Cropp wrote: > I work on the Asterisk side of things and admit to not knowing about > browser development.

Re: [asterisk-users] Webrtc and iOS devices

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 wrote: > Hello, >

Re: [asterisk-users] Webrtc and iOS devices

it working on most recent version of Asterisk On Tue, Apr 28, 2020 at 11:37 AM Teijo wrote: > Hello, > > > Currently audio conference. Should upgrading Asterisk from 13 to newer > version resolve webrtc/iOS problem? > > > Best regards, > > > Teijo > > Da

Re: [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ?

son that couldnt go out to a radio station stream for example.. On Wed, May 6, 2020 at 8:55 PM Jonathan H wrote: > 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 > > O

Re: [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ?

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, wrote: > Way back in 2016 the only way to allow callers to listen in to a stream > "at will" was to do the

Re: [asterisk-users] Max calls per box

I've heard of people having thousands of channels on a box Dovid. Not how I would personally do it myself but if you've already got the hardware. And think about if you can simplify the deployment by accessing the sound files via http so you only have them in one place On Thu, 19 Mar 2020,

Re: [asterisk-users] Stir Shaken is upon us

. Colp wrote: > On Sun, Jul 12, 2020 at 11:37 PM Michael Maier > wrote: > >> On 13.07.20 at 00:17 Joshua C. Colp wrote: >> > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins wrote: >> > >> >> Asterisk 18 will have support based on this asterisk update Ma

Re: [asterisk-users] Stir Shaken is upon us

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, wrote: > On Sun, 12 Jul 2020, Saint Michael wrote: > > > WORLDWIDE EMERGENCY > > Again? > > > The code below needs to be

Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem

As far as I'm aware Josh, it doesnt stop a call from happening - I've had the same "errors" pop up when using Twilio and Simwood but calls continue just fine. On Thu, Dec 2, 2021 at 2:30 PM Joshua C. Colp wrote: > On Thu, Dec 2, 2021 at 10:18 AM James Cloos wrote: > >> > "KT" == Kingsley

Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem

It shouldnt stop the call from happening. It will be something else... up your debugging level and see what else you get Lots of providers go against this part of the spec but I've run Asterisk 18 with twilio over sip over tls and everything worked, it just spat out the error line On Thu, Dec