search for: userev

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-...

...e 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 the call is being placed from. Is there anything stupid that I may have done, that will prevent asterisk from issuing the above call when the call is hung up?

...or not, who has forwarded his call to his voicemail, etc ... When using manager's login command with Event parameter set to on, I'm getting tens of events I don't care about but I suppose I won't miss things like transfers, pickups, parking ... Would it be a right move to rely on UserEvents instead ? Then I would specifically have to add those UserEvents in dialplan but I'm afraid to be unable to support things like hangups or transfers, ... What's your opinion about that ? Would you "filter system events" or "add custom uservents" ? Regards ---------...

...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 (NOANSWER,BUSY,CHANUNAVAIL ,CONGESTION,ANSWER) exten => s-CHANUNAVAIL,1,NoOp("CHANUNAVAIL") exten => s-CHANUNAVAIL,2,UserEvent(ChannelUnavailable|Account: ${ACCOUNTCODE}^${CALLERIDNUM}) exten => s-CONGESTION,1,NoOp("CONGESTION") exten => s-CONGESTION,2,UserEvent(Congestion|Account: ${ACCOUNTCODE}^${CALLERIDNUM}) exten => s-ANSWER,1,NoOp("ANSWER") exten => s-ANSWER,2,UserEvent(Answer|Acc...

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. I...

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 popups and in the use of UserEvent...

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

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 process SIP MESSAGE requ...

...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 (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-CHANUNAVAIL,1,NoOp("CHANUNAVAIL") exten => s-CHANUNAVAIL,2,UserEvent(ChannelUnavailable|Account: ${ACCOUNTCODE}^${CALLERIDNUM}) exten => s-CONGESTION,1,NoOp("CONGESTION") exten => s-CONGESTION,2,UserEvent(Congestion|Account: ${ACCOUNTCODE}^${CALLERIDNUM}) exten => s-ANSWER,1,NoOp("ANSWER") exten => s-ANSWER,2,UserEvent(Answer|Acc...

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 => _X.,1,UserEvent(TAPI|TAPIEVENT: [~${UNIQUEID}] LINE_CALLSTATE LINECALLSTATE_CONNECTED) exten => _X.,2,Dial(SIP/${EXTEN}@MACHINE <mailto:SIP/${EXTEN}@MACHINE>) exten => _X....

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

...configurations. I'm hoping somebody here has an answer. Currently, I have this in extensions.conf [app-dnd-on] exten => *78,1,Answer exten => *78,n,NoOp(${CALLERID(num)} channel ${CHANNEL} is going on DND ACTIVE) exten => *78,n,Set(DB(DND/${CALLERID(num)})=On) exten => *78,n,UserEvent(ASTDB|Family: DND^Channel: ${CHANNEL} ^Value: On) exten => *78,n,Playback(do-not-disturb&activated) exten => *78,n,Hangup [app-dnd-off] exten => *79,1,Answer exten => *79,n,NoOp(${CALLERID(num)} is going OFF DND) exten => *79,n,DBdel(DND/${CALLERID(num)}) exten => *79,n,...

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 Sangoma Technolog...

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

I want to add to my dialplan the ability to spy on an arbitrary extension whether a call originates at it or is terminated at it. Scenario 1: Given an extension, say 2001, a call comes in on a zap channel and is Dial()ed to the phone that's at extension 2001, I want to be able to pick up a phone and dial (say) *142001 and spy on that call. Scenario 2: Extension 2001 makes a call to, say a

...ERROR) exten => 7080,n(TIMEOUT),Goto(ERROR) exten => 7080,n(JOINUNAVAIL),Goto(ERROR) exten => 7080,n(LEAVEUNAVAIL),Goto(ERROR) exten => 7080,n(JOINEMPTY),Goto(ERROR) exten => 7080,n(TIMEOUT),Goto(ERROR) exten => *210,1,AddQueueMember(teste,SIP/${CALLERID(num)}) exten => *210,n,UserEvent(RefreshQueue) exten => *210,n,Playback(agent-loginok) exten => *220,1,RemoveQueueMember(teste,SIP/${CALLERID(num)}) exten => *220,n,UserEvent(RefreshQueue) exten => *220,n,Playback(agent-loggedoff) In queues.conf: [teste] strategy=roundrobin music=default timeout=10 retry=0 maxl...

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 to setup a "cti server" that talks with both AMI and AGI channels, the first one mainly when one just want to send some data from asterisk to the "cti server", and the se...

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", "") in new stack == Spawn extension (heartbeat, s, 2...

Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel