similar to: Sent ami event from AGI?

Is it possible to simply send a message to appear as an AMI message/event, from the dialplan? For example exten =>123,1,ami(myEvent, param1, param2) and in the AMI a message appears like: Event: myEvent Privilege: call,all Channel: PJSIP/misspiggy-00000001 Uniqueid: 1368479157.3 ChannelState: 3 ChannelStateDesc: Up CallerIDNum: 657-5309 CallerIDName: Miss Piggy

Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > > Can anyone tell me how can I create echo test using ARI stasis > application?

I'm pretty sure there isn't a way to do that currently. ?My best guess would be that a new special type of bridge technology could be created that would implement the per-channel echo (no audio bridged between channels in the bridge). That would require new C code in Asterisk for the bridge, and then the usual methods of moving channels in to bridges with ARI could be used.? On Sat, May

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

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

We are trying to add voice mail to our hotel rooms. Our current phone instruction cards say 'to reach voice mail dial ext 456". Replacing those instructions is not feasible at the moment. We have Feature Code *97 that takes them directly to their voice mail box. Question - What is an easy way to have exten 456 dial *97. We are using AsteriskNow distro, version11. Phil Ledon

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 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

Hello, I'm using an AMI Originate command to send a fax. The fax is sent by a script, and I'd like my script to send the fax, wait until it has succeeded or failed, then exit with an appropriate error code (it is driven by a mail system, so the exit code will tell the mail system whether to retry the fax later). The script works great if the fax succeeds, or if the line is busy or

Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: binary <dreamer.binary at gmail.com> </div><div>Date:01/07/2014 6:09 PM

Is it possible to detect the failure of an agent to register with Asterisk via the AMI ? When I try to register with Asterisk 1.4 using an invalid password I don't see any event in the AMI, but see this in the messages log: [2013-10-05 22:05:03] NOTICE[24598] chan_sip.c: Registration from '"test"<sip:230 at asterisk.com>' failed for '192.168.0.1' - Wrong

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

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

HiCould someone please help me how to set Custom generated UUID in Originate action in AMI ? RegardsBabak -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150508/528d5ff1/attachment.html>

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

Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table ?direct_media? sets to ?no? on all endpoints but it doesn?t help. if native_rtp not work for some reason I have oneway audio. how can I fix this?

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

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries