similar to: [asterisk-app-dev] ARI Node JS Bridge.addChannel

Hi, I'm writing a node.js backend to pass events via a websocket to a CRM. Basically what I want to do is notice when things happen (i.e. new channel, new bridge etc) without sending the channels to the Stasis app. The channels I'm interested in are agents who are in a queue only because they are in a realtime MySQL database for the queue_member_table. There doesn't appear to be a

Hiya, When I hang up on a call to my stasis app I’m getting multiple channelDestroyed events for the same channel: app.js:985:13) Channel was destroyed: 1547220509.77 app.js:1029:17) This was a customer app.js:1030:17) Checking if this was a customer talking to an agent app.js:1043:21) Customer was not talking to anyone app.js:1126:13) 2019-01-11 10:28:29 app.js:985:13) Channel was destroyed:

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

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

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

Hi there! Is there any way I can turn on debug for ARI and sending the output to a separate log file? So far I have only been able to turn on ARI debugging in the console which results in the debug output being logged in /var/log/asterisk/messages I would love to have ARI debug log messages in /var/log/asterisk/debug or even better in it's own ari-debug file. With best regards Florian

Up till now, I have only used Asterisk versions 1.2, 10 and 11, on CentOS 4, 5 and 6, and have made extensive use of AMI and FastAGI connections to a multi-threaded backend written in C. For a new project, I am looking at trying Asterisk 16 with ARI, on CentOS 7. I was looking at the various ARI libraries available, particularly the ones for Python and Node.js in github. I noticed that the

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

On 6/26/23 9:00 AM, Joshua C. Colp wrote: > 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

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

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

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

We are looking to migrate from AMI to ARI. We currently rely heavily on ConfBridges for multiple party support. Is it possible to add more than 2 channels? If so, is there a limit? Or a way to configure the limit? Have a great day! Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm attempting to run a test of the ARI recording of audio from the channel. When I send the record command, it's failing. curl -v -u asterisk:asterisk -X POST "http://locahost:8088/ari/channels/mychanntest.1/record?name=mytest&format=WAV&maxDurationSeconds=300&maxSilenceSeconds=3" [08/11 09:14:13.290] WARNING[23806]: ari/resource_channels.c:812

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

Hello, I am experimenting with Asterisk 12.0.0 alpha1. I have a couple of SIP phones working. Good. I can retrieve data using curl to interact with the new Asterisk REST API (ARI). Good. Now I want to use the new ARI events API, which requires a WebSocket connection. I am using Node.js for the client, and have a stable connection to ARI events on the Asterisk 12 server. What I hope for is

Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a number is being played? Here is a test I am running. I am playing multiple portions (sounds and numbers). curl -v -u asterisk:asterisk -X POST http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,sound:demo-instruct,number:123,number:456,number:789 If I attempt to

On 6/26/23 5:19 PM, Jeff LaCoursiere wrote: > On 6/26/23 9:00 AM, Joshua C. Colp wrote: >> 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

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 to read these from either the ARI or AMI ? I'm trying not to modify the dialplan. Thanks Brian

ARI (Asterisk Recording Interface) has reached another milestone. The project is starting to become a full featured user portal and handle all the common errors that people seem to have. This release supports: call monitor page ? new features include column sorting and filter small duration calls in addition to the ability to listen to call monitor