similar to: ARI events : ChannelDestroyed and ChannelHangupRequest

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:

On Wed, Jun 7, 2023 at 10:46 AM TTT <lists at telium.io> wrote: > I’ve reread the documentation a few times, and what isn’t clear is whether > I need an app=X parameter in the url. In other words, can I only get > events for a single named statis app? Or can I get events for the entire > Asterisk server? > > > > The command below (without app= parameter) results in

I’ve reread the documentation a few times, and what isn’t clear is whether I need an app=X parameter in the url. In other words, can I only get events for a single named statis app? Or can I get events for the entire Asterisk server? The command below (without app= parameter) results in no events being shown, but no error either. Thanks Brian (Ast newbie) From: asterisk-users

Sorry, I forgot to write that the SIP peer must keep ringing while the announcement is being played. Le 22/08/2016 ? 17:42, John Kiniston a ?crit : > This seems like the obvious answer but maybe I'm misunderstanding the > question. > > exten => s,1,Dial(SIP/alice,20) > same => n,Playback(myannouncement) > same => n,NoOP(Whatever else you want to do goes

Ok that worked. Since I have not declared a statis app called “test”, does that mean any non-existent app name on the URL will subscribe to all system events? (Or is test a built-in app name) Brian From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Wednesday, June 7, 2023 9:56 AM To: Asterisk Users Mailing List - Non-Commercial

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. >

How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer

Hello, I am searching a way to dial a SIP peer, and if it does not answer within 20 seconds, play an announcement to the caller. This means that the caller would hear a ring tone for 20 seconds, and only then hear the announcement if the callee did not answer. I know it is possible to do this with ARI, but in this particular case I do not want to use ARI. I would like to do this purely with

On Thu, Jan 30, 2020 at 12:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote: > Hello, > > I use UserEvents generated by the Message/ast_message_queue channel with > the UserEvent application. > > Regards > > Jean > Thanks Jean. We're looking at alternatives. > Le 29/01/2020 à 20:31, George Joseph a écrit : > > For those of you who actually

Maybe try progress() instead of answer () ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If the SIP peer is available, I only get > the ring tone, and never hear

On Thu, Jan 30, 2020 at 3:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote: > Hello, > > I use UserEvents generated by the Message/ast_message_queue channel with > the UserEvent application. > Do you use any aspects of the channel itself in the user events, or merely the contents of the user event and what you've placed in it? -- Joshua C. Colp Asterisk Technical Lead

Hello, Has anyone already implemented some sort of call preemption in Asterisk ? I am trying to achieve something like this : - I want to limit the number of calls on a given SIP peer to 10 - on the other hand, some calls have higher priority than others - when the ceiling of 10 calls is reached and a call with a high priority is attempted, I would like to drop a call with a lower priority

Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO received **on **SIP/xxx-00000004:* [Dec 13 11:56:16] DTMF[18193][C-00000005]

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

For those of you who actually process SIP MESSAGE requests... Do you use any of the AMI events generated by the "Message/ast_msg_queue" channel? We want to change that channel to an "internal" channel that doesn't generate AMI events (for performance reasons) but we need to know if anyone's using them first. Thanks! -- George Joseph Asterisk Software Developer

On 2020-01-30 10:30 a.m., George Joseph wrote: > > > On Thu, Jan 30, 2020 at 12:18 AM Jean Aunis <jean.aunis at prescom.fr > <mailto:jean.aunis at prescom.fr>> wrote: > > Hello, > > I use UserEvents generated by the Message/ast_message_queue > channel with the UserEvent application. > > Regards > > Jean > > > Thanks

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]

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

On Wed, Jun 7, 2023 at 12:04 PM TTT <lists at telium.io> wrote: > Ok that worked. > > > > Since I have not declared a statis app called “test”, does that mean any > non-existent app name on the URL will subscribe to all system events? (Or > is test a built-in app name) > Applications are not declared or configured anywhere. The act of connecting a websocket with an

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