similar to: Subscribe to events via ARI from node.js without sending to Stasis

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation). ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com> Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re:

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

On Thu, Aug 6, 2020 at 1:52 PM Dan Cropp <dan at amtelco.com> wrote: > I understand how to control the first local channel, but an having trouble > getting the second local channel to enter stasis. > > > > I setup have the following extensions.conf to handle 1000 (basically had > it setup so if first stasis not there try second, but believe second > channel never

Le 2015-12-15 15:25, Joshua Colp a ?crit : > Sylvain Boily wrote: >> Hello, > > Just a note - there's an asterisk-app-dev mailing list[1] which is > better suited for these kind of posts. Ok > >> I did some tests because i'm interesting to transfer a non stasis bridge >> to a stasis bridge and i found a strange situation. > > You can't, you have

Hello, I did some tests because i'm interesting to transfer a non stasis bridge to a stasis bridge and i found a strange situation. A call B B answer You have a bridge On my asterisk CLI: xivo*CLI> bridge show b1d8fb21-ec6d-469a-9dde-bb6bfd5618cc Id: b1d8fb21-ec6d-469a-9dde-bb6bfd5618cc Type: basic Technology: simple_bridge Num-Channels: 2 Channel: SIP/tcu9tz-00000032 Channel:

Hello. I tryto use Statis at my dialplan to run my app (a) When Statis is running from making call ( I dial from softphone some exten and run dialplan context where call Statis(MyApp)) Asterisk responsed: ERROR[61517][C-00000019]: res_stasis.c:852 stasis_app_exec: Stasis app 'MyApp' not registered How I must Register MyApp -------------- next part -------------- An HTML attachment was

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

Can anyone tell me how can I create echo test using ARI stasis application?

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

I have the ARI enabled on my Asterisk test box, and want to listen to all events. I can't find the syntax to do that. Can I only listen to events related to a stasis app? I was hoping that a simple wscat command like this would show me all events: wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk " I know how to do it form the AMI.looking for

hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek

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

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

I tried the command below (with subscribeAll=yes). I made a couple of calls but didn’t see any events. Should I see events? From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Tuesday, June 6, 2023 6:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users]

On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > I am connecting to the ARI with subscribe all, so I can see channels being > created. I now want to extract a variety of header variables (at the > moment the from and to tag). I tried to read them from the ARI but > Asterisk refuses since the channel is not in a stasis app. > > > > Is there a way

I tried GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id) But it responds with "message": "Channel not in Stasis application" Since I want to get the call-id for a channel not in stasis I guess that won’t work. Similarly, I can’t force the channel through my own code in the dialplan, so the PJSIP_HEADER function won’t work. So it looks like I’ll

this code worked for me, here is what I did and worked for me: exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) Thanks for you help! On 04/27/2015 02:56 PM, Matt Riddell wrote: > >> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com >> <mailto:motty.cruz at gmail.com>> wrote:

> On 18Jun, 2015, at 13:44, Greg Woods <greg at gregandeva.net> wrote: > > I am certain that the old number that is showing up as my caller ID is not present in any of my config files (that includes sip.conf and iax.conf, everything in the /etc/asterisk directory has been checked). Did you buy the number from your carrier? Maybe it?s set on their side for the trunk. -- Cheers,