similar to: AGI and exec Playback

On Wed, Jun 6, 2018 at 6:18 AM, Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote: > > > Hi, > > > > I have tested ControlPlayback and grabbed files via an apache server with > > no issue. > > ControlPlayback is an Asterisk dialplan function. > > How have you integrated this

Humm... Seems like my message didn't make it. Here we go again.. /Johan -------- Original Message -------- Subject: Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer <lists at jttech.se> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> On 2011-06-05 19:54, virendra

Hi List, Asterisk 's *ControlPlayback* will used for play any recorded file as an audio player. Is it possible that we can use it for multiple forward and rewind ? ex:- original: ControlPlayback(filename,skipms,ff,rew,stop,pause) expected ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause) : ----- Thanks and regards Virendra Bhati

There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks

Hi, I have tested ControlPlayback and grabbed files via an apache server with no issue. I want to be able to grab files via aws S3 which would require me to add some headers to authenticate. Is there any way to have Asterisk add headers or would I need a http proxy in the middle? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

Asterisk People, I'm currently using Asterisk and with a SIP voip provider and I'm having problems where DTMF input in my IVR app is getting corrupted intermittently. For example, if someone enters 1025, it may come though correctly as 1025, or it may come trough as 10025, or 100255. DTMF digits will just double up. This doesn't happen all the time. Asterisk will just pick times

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

If not sure if I am looking at a bug or expected behaviour as I do not see anything in the documentation. ControlPlayback can not replay complicated file names For example it can replay 1005 but it can not replay 1005-2014-04-08_23:58:17 Playback can replay 1005-2014-04-08_23:58:17 I suspect this relates to how the variables are parsed and parameters set. Does anyone have any further

Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor

Hello! I have developed an IVR using AGI and so far it works great. I'm using Cepstral voices, but now want to use the voices from AT & T that are on a Windows server to be heard best. With cepstral what I do is to generate audio files from shipping and this text I reproduce this method it has worked very well. Now, try to do the same by creating the audio file in windows with the

Hi, I'm currently migrating to 1.4 and have problems changing deprecated AgentCallbackLogin to AddQueueMember. I have dynamic queues and dynamic agents (MySQL Realtime), and pseudo-dynamic agents.conf (with huge amount of possible agent numbers). Agent login is done trough manager API: * AgentCallbackLogin * QueueAdd In 1.4 seems AddQueueMember can do all the same, but there is no such

Hi, We've run into an interesting (to us) problem with SpeechBackground. Inside a AGI script, we're playing some extended audio?basically, like a podcast?and we want playback to stop if and only if the speech recognized matches something in our grammar. If there's speech that doesn't match, we just want to go right on playing. (We're using LumenVox as our speech recognition

There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of

Dear all, I want playback or ControlPlayback cmd to trigger me when a DTMF key is pressed, so I can execute Monitor cmd or any thing I want. Anyone did this job before?. Please help me. Thanks in advance, Giang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100324/0884f55a/attachment.htm

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk >From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can

I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following: exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70) There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the

Hi. I've been using the ControlPlayback function as part of an IVR system, but am finding it very restrictive. Is there any way to tell it to play hold music while the user has pause selected? I don't want the line to just go silent indefinitely. If I want the caller to have a pause option, is there some alternative to using ControlPlayback? I think I've got the hang of doing fancy

This 2 line code is doing what I wanted. exten => 200,1,voicemail(200) exten => 200,2,Hangup What I've been told is that they want the 20 year old phone system to light up the message bulb. (yea, a filament bulb, not an LED) To do this you pick up on the line that goes into Asterisk and do a: exten => 200,1,SendDTMF(200w#86) But I don't know the path to take to get that

Hi. I'm new here so I hope this is a sensible question/sensible place for it. I have a PSTN to IAX phone number with voipuser.org that I'm using to test an IVR service. The only trouble is that after approximately 40 seconds of silence (e.g. after background playback of a menu prompt) the call gets cut off. Is this a common problem? I've already set the ResponseTimeout to a big