similar to: Should you use UserEvents for monitoring calls ?

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

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as

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 all, I try to install FOP. It's very nice. In documentation I red that from my dial plan I can launch a popup window with UserEvent() application. I try to follow FOP documentation but I can't popup anything. My structure is: - server 1: Asterisk system - server 2: FOP system - client On client I connect to FOP panel, but I don't see any popup. Someone can help me to configure FOP

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 tried to use UserEvent() command to send data to Asterisk Manager from my dialplan. It works fine if the body only contains 1 line but I don't know how to send multiple arguments in the body. I have tested: UserEvent(eventname|body1|body2) UserEvent(eventname|body1\r\nbody2) But no one seems to work. Is it possible to do that and what is the correct syntax? Amaury

Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status

Hi, I need a hack to query current calls coming in and going out an Asterisk 1.6.1 system and send this list back as a custom UserEvent to other listening endpoints. For various reasons, I might need to write this hack in PHP though I'm more experienced with Asterisk Java. What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ? I'm referring here to

Hi, I have been playing around with the latest release of asttapi and I have found the 'hangup' problem already reported to the list here <http://lists.digium.com/pipermail/asterisk-users/2006-May/151260.html> Apparently hangup should be done by making use of UserEvent commands. So I have configured this context for being used when making calls from outlook: [outlook] exten =>

I have written an asterisk manager client which creates an outbound call using Asterisk manager API's Originate action. when the call is connected I run 3 applications on it. 1)read a dtmf digit from user 2)A customized application which I have written,(It plays something to user) 3)Hangup If user hangs up while app 2(see above) is executing I get a 'Event Hangup' from asterisk in my

Hello, Some of my users have phones lacking a DND button. I need to provide an extension they can dial that will put them in DND, i.e. tell the server not to send them any calls until they get off the DND. I've researched it for almost 3 days now and tried a range of configurations. I'm hoping somebody here has an answer. Currently, I have this in extensions.conf [app-dnd-on]

Hello, I'm using 1.6.1-rc4. When sending a userevent, (with "UserEvent(MyEvent);" in extensions.ael), I've got : Event: UserEvent Privilege: user,all UserEvent: MyEvent I can't see any Uniqueid field as mentioned http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent or http://www.the-asterisk-book.com/unstable/applikationen-userevent.html Is this Uniqueid mandatory ?

Hello everyone. I got a little problem in here: I want to set up a queue so that if anything of these happens: a) No agents logged in b) All agents busy Then the user gets diverted somewhere. I used this (for testing purposes only, of course): exten => 7080,1,Answer() exten => 7080,n,Queue(teste) exten => 7080,n,Goto(${QUEUESTATUS}) exten => 7080,n(ERROR),NoOp(${QUEUESTATUS}) exten

Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:3333/fetch,queuename) The idea is

I am running a heartbeat... Asterisk 11.15.0 - same behaviour is noticed on 1.4.43 also I issue a call through the API that does the below. just UserEvent and Hangup -- Executing [s at heartbeat:1] UserEvent("Local/s at heartbeat-0000000f;2", "HeartBeat, Noop") in new stack -- Executing [s at heartbeat:2] Hangup("Local/s at heartbeat-0000000f;2",

Hi, I am experimenting with getting hold of the sip cause and sip response from outgoing call. How could i make a userevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise -

Hello, I use UserEvents generated by the Message/ast_message_queue channel with the UserEvent application. Regards Jean Le 29/01/2020 à 20:31, George Joseph a écrit : > 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

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

Hello User-List, is it possible to play a sound file directly to a caller channel? Like this AMI command Action: Originate Channel: SIP/20-00001d41 Application: Playback Data: /path/to/audio/file I get an Error Message. My intension is to play a sound file to a caller and the other callers don't hear this. Can someone help me ? Thanks a lot Bye Daniel

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where