similar to: Fwd: Re: ControlPlayback's options

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

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent "issues" that

Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1- Based on above error and Google search I have

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

Hi, I try to find some information about CAS E1 signalling and how it's handled by Asterisk. My customer wants to connect to a BT ITS Netrix by CAS E1 E&M. The system is intended to take the channels and mix them (meetme / confbridge) and send the audio back mixed to each. The layout: BT ITS Netrix: CAS E1 E&M <-> MUX - WAN - MUX <-> Digium TE220, Asterisk I've

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

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ errors when running "dahdi show status". I see that sharing IRQs for Digium cards isn't

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:

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

Hello, I'm trying to cut off the "+" sign if part of a telephone number, but not succeeding : exten => test,n,Set(cid=+99999600) exten => test,n,Set(regx="([0-9])") exten => test,n,Set(cid2=$["${cid}" : ${regx}]) exten => test,n,NoOp(cid2=${cid2}) cid2 is empty afterwards... What I want is to make sure there are only numbers and no other

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, I write some php code in AMI to working with asterisk command. I don't know exactly what is the different between AMI and AGI and witch one is better for my planning. Im planning to call party users that their number is is my panel on web. We have some operator and they can call party users via client softphone by clicking on their number, so they have to limited to call just listed

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, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading "User Profile Configuration Options" the option announce_only_user is present. The sample config looks like this: -- ;announce_only_user=yes ;Sets if the only user announcement should be played when a channel enters a empty

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

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

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

On 04/07/2015 10:48 AM, Johan Wilfer wrote: > Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: >> Dear all, >> >> Is anyone has experience making Asterisk server with virtual server >> OPEN-VZ (in proxmox 3.4 box) ? >> >> My boss want to build a production server with it, and it will have +/- >> 300 sip user (concurrent call maybe < 150 call) >>