similar to: [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

Hello, I noticed that when a channel is destroyed, two different events can be raised : ChannelDestroyed and ChannelHangupRequest. These two events seem to be mutually exclusive : if I receive a ChannelHangupRequest, I will never receive a ChannelDestroyed, and vice versa. This behaviour does not look consistent with the documentation, which states : "ChannelDestroyed : Notification

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

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

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

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

Hi there, I have a system on which btrfsck gives the following output... I don''t understand the meaning of the reported errors, so any clue would be appreciated. Is this something I should worry about, or not ? Would I be advised to try "--repair" ? (Last time I tried this one, it completely b0rked a filesystem, beyond repair, and my wife would kill me ifever I trash this one,

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

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

Hi, I have a function that approximates some data and indicates "segments". I'd like to plot the original data, and then the linear approximations on top of it. (Ideally, just a subset of N rows at a time, as the data set is large.) I can't figure out a clean way to do this. Suggestions? here is some sample data: ================================== Row X Seg ?.

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

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

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. >

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

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

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

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 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]

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

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

2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk