similar to: Hook playback or ControlPlayBack cmd

Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML

Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk

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

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

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, 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:

Hello, I'm looking for a way to play sound file, and control the playback trough web interface. Is it possible to use AGI to play a sound file and then by receiving some event stop playing it, and play another file. The catch is that i want to seek to 1st minute, 5th minute, etc - so regular ControlPlayback with intervals wouldn't fit - i have to use sox to create different file and then

Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load

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

One thing that concerns me is that this post is from 2009, even though the newest version of the app on Sourceforge is up to date. I have a customer who has been using a conference server that I built for him using app_konference for several years now and he routinely runs conferences with anywhere from 10 ? 125 active users. The ultimate goal is several hundred concurrent users and I can see that

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

Using Asterisk 1.4.31 and addons 1.4.11, ControlPlayback get confused when skipping forwards on an mp3 file (it seems to work fine on wav's). I'm calling it from an AGI like so: $agi->exec('ControlPlayback',$filename . "|4000|#|*|8|0|7"); The first four times I press the '#' key it does indeed skip forwards; but the fifth and subsequent times pressing

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 an extension to use ControlPlayback -- however the forward and rewind controls do not work if I am in a 3-way call with the pstn and that extension, however if I then hang up the controls work. Anything to do about this? Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici

Is there a standard set of prompts to use with ControlPlayback as there are with many of the other applications like confbridge? I can?t find a prompt with instructs the user how to use it?s functions. Perhaps you have to construct a prompt out of multiple prompts. Does anyone have an idea of which ones to use? Regards --- This email is free from viruses and malware because avast! Antivirus

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 all, I'm being requested to deploy an?IVR service?using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, How can I define an array of sip number in sip.conf ? I want to define an array of sip number from 1000 to 2000, so I can make a performance test on Asterisk using sipp. Thanks in Advance, Giangnh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100318/858dcd1d/attachment.htm

Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk can process much more call at a time than with faster speed it is not caching. Thanks, -------------- next part -------------- An HTML attachment was

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