similar to: New mailing list - asterisk-app-dev

recreate Echo, if that is possible. trying to recode all dialplan to stasis application > On 22 May 2015, at 19:29, Scott Griepentrog <sgriepentrog at digium.com> wrote: > > Nick- > > Are you wanting to recreate the dialplan Echo() application in stasis? > > Why not just send the call to Echo() instead of Stasis()? > > On Fri, May 22, 2015 at 11:25 AM, Matthew

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

I'm pretty sure there isn't a way to do that currently. ?My best guess would be that a new special type of bridge technology could be created that would implement the per-channel echo (no audio bridged between channels in the bridge). That would require new C code in Asterisk for the bridge, and then the usual methods of moving channels in to bridges with ARI could be used.? On Sat, May

Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > > Can anyone tell me how can I create echo test using ARI stasis > application?

On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote: > > I get the desired use case to run app_amd from within a Stasis > > application, but I’m not sure about app_queue. You have everything at > > your disposal within ARI itself to replicate all of the functionality > > of app_queue and

On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > Can anyone tell me how can I create echo test using ARI stasis application? > I'm not sure an 'echo' test really makes much sense with ARI, but we do have some nice documentation on getting started with ARI on the wiki. The basic tutorial example should give you an ARI event over a WebSocket

On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: > Paul Albrecht wrote: >> Really? Shouldn?t something this major affecting the entire Asterisk >> community get discussed on the lists? Any idea what Leif is talking >> about when he says the community is in transition, moving from dial >> plan model to external control. > > It was

On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk <62mkv at mail.ru> wrote: > Hi all > > Have recently watched Matt Jordan's session on Kamailio World 2014 > > On slides 26-29 of his presentation > (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) > he speaks about a (completely new, for me at least) approach to build

Hey, trying to use ARI with NodeJS - this doesn't work: play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav'); should it? https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Simple+Media+Manipulation says: A sound file located on the Asterisk system. You can use the /sounds resource to query for available sounds on the system. You can also use specify a media

Join a worldwide community of designers, developers, and communications technologists to to create, code, and design apps built on Asterisk and other communications APIs at the inaugural AstriCon Hackathon! Obligatory yet useful information: *When*: Wednesday, October 22nd at 8AM. A reception is on Tuesday, October 21st from 5 - 7 PM. *Where*: Red Rocks Casino in Las Vegas, Nevada (at AstriCon!)

Each year at AstriCon, we have an Open Source Pavilion which showcases projects which are adjuncts to Asterisk, or which are directly relevant to improving the utility and features of Asterisk. It gives smaller projects the chance to have some space to display what they're doing, and to give demonstrations to the AstriCon attendees who are interested in some of the things happening

2013/9/19 Matthew Jordan <mjordan at digium.com> > > On Thu, Sep 19, 2013 at 9:02 AM, Olivier <oza_4h07 at yahoo.fr> wrote: > >> Hi, >> >> Asterisk 11 doc says CDR(src) value is read-only (see >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). >> >> For various reasons, I would appreciate to change its value so that it

2014-08-13 19:06 GMT+02:00 Matthew Jordan <mjordan at digium.com>: > On Wed, Aug 13, 2014 at 12:01 PM, Olivier <oza.4h07 at gmail.com> wrote: >> After installing various packages, here is what I did: >> >> TDIR=/usr/src >> cd $TDIR >> PJOPTIONS="--prefix=/usr --enable-shared --disable-sound >> --disable-resample --disable-video

Hi Asterisk users, I'm one of Asterisk ARI users, and trying to designing the new ARI for application execution in Stasis(). This will be made possible for executing the applications in the Stasis() application. But, before going further, I would like to know which application needs to be considered. Because this feature will introduce new Stasis behavior, I would like to test the

On Oct 22, 2014, at 11:47 AM, BJ Weschke <bweschke at btwtech.com> wrote: > On 10/22/14, 12:14 PM, Paul Albrecht wrote: >> On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: >> >>> Paul Albrecht wrote: >>>> Really? Shouldn?t something this major affecting the entire Asterisk >>>> community get discussed on the lists?

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP providers are not very good at doing that suppression (leaving audible clicks, or failing to suppress the tones

Hiya, I would have expected this to show the channels in the bridge inside the anonymous function - it shows the bridge is empty though? var bridge = ari.Bridge(); bridge.create({ type: 'holding', name: event.application+" bridge" },

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries

Greetings, Asterisk 1.6.2.0 was released last week. It's time to revisit release plans for both current and future Asterisk releases. For the past few months, there have been discussions regarding some updates to Asterisk release policies. You can find my original -dev list post on this topic here [1]. I also spoke about this as part of my presentation at AstriCon, which you can find

Matt Riddell wrote: > >> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com >> <mailto:luca.pradovera at gmail.com>> wrote: >> >> I have been working on designs for two different projects, where both >> of them would need to use the IBM Watson streaming ASR service. >> >> Would it be possible to write out the audio frames