similar to: play a sound file directly to a caller channel

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

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with:

Please don't top post. > Am 29. Mai 2015 09:42:55 MESZ, schrieb Luca Bertoncello > <lucabert at lucabert.de>: >> Zitat von jg <webaccounts173 at jgoettgens.de>: >>> Yes, it is called "core set verbose 42", the other options is "core >>> set debug 42". Enjoy the show! I know you can specify a level to the verbose application,

hi list, i have a question about the args of queue: when we use Queue() app, there are some arguments than can use. help from CLI: Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]]) well.. i'm trying to identify who has taken the call on a queue, and, when agent conected, launch a macro with some args based on who takes the call i do: exten =>

I have an extension defined like this: exten => do_monitor,1,Answer() exten => do_monitor,n,NoOp(Just got '${CfMC_ActionID}') exten => do_monitor,n,ChanSpy(${CfMC_WhoHear},qX) exten => do_monitor,n,Hangup() I use an AMI packet like this: Action: Originate Channel: Agent/1001 Exten: do_monitor Context: cfmc_cdi_private Priority: 1 Variable: CfMC_ActionID=callE1334 Variable:

Hello, I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup: Reason: q.850;cause=17 Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100221/d29c02b8/attachment.htm

I would like to be able to stop the chanspy application and go to the next step in the dialplan but I do not see a way to do that. I have looked at the code and I do not see a way to stop the chanspy application. Even if there are no channels that match the chanprefix pattern the chanspy application is not exited. Hitting the * key stops spying on a channel but then starts spying on the same

Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

Hello guys, I'm using asterisk 11. i'm using Chanspy in a local channel to playBack a file to a specific channel. [playsound] exten => do_playback,1,Answer() same => n,Wait(1) same => n,Playback(${Pv_WhatToPlay}) same => n,Hangup() exten => do_chanspy,1,Answer() same => n,ChanSpy(${Pv_WhoHear},qXBwW) same => n,Hangup() just basically

You're right Steve, sorry for that. So Hi again guys. I need a little help here.I'm using asterisk 11. i'm using Chanspy in a local channel to playBack a file to a specific channel. [playsound] exten => do_playback,1,Answer() same => n,Wait(1) same => n,Playback(${Pv_WhatToPlay}) same => n,Hangup() exten => do_chanspy,1,Answer() same =>

Hi List; I am trying to use wiki via the link (http://www.voip-info.org/wiki/index.php?page=Asterisk) in effective way to find the needed resource for me, but still it is hard to arrive for the needed information. For example: what is the best (shortest) way to search for information related to the command playbak()? Using the backlines, it make the eyes feel hard by keep reading without

I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 10 seconds Asterisk exits the RetryDial step with HANGUPCAUSE=0 and

HI Matt, I am trying to replicate this particular problem. We are seeing more frequently where the Event: AsyncAGIExec is never being sent. The two scenarios I have seen in tests yesterday and today... We sendl an AMI action. For example, play a short file or hangup. AMI Events will indicate it did the work, but we never receive the Event: AsyncAGIExec with a result at all. Asterisk debug

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

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