similar to: Dial(L(x...)) distinct to SET TIMEOUT(absolute)

well, I took a look at the files. from my knowledge of shorten there are two things it does that flac doesn't do: 1. it estimates the mean of the signal for each block, subtracts it out and stores it separately. but this is pretty useless for the predictors that shorten uses as they are pretty insensitive to the mean (try different values of -m from 0 to whatever and note practically no

Hi Folks, I dont know how exactly to start... so im going to (what i think is) the point... In a dialplan, after i set an autohangup (with AGI) , how could i send a sound (stream a sound ) into an open channel at X seconds before the autohangup time get to 0 for that channel? (Like public phones, that gives u a 'beep!!!' before ur time runs out, just like that...) Thank you very

On Thu, 15 Feb 2001, Josh Coalson wrote: > well, I took a look at the files. from my > knowledge of shorten there are two things it does > that flac doesn't do: > > BUT, you have stumbled on some recordings where the > LSB is 0 for much of the file. as a matter of fact, > in > the worse track (track 6) almost the entire signal in > both channels is like that.

Hi All, suppose this: Dial(SIP/<somecarrier>/<somenumber>/60/L(3600000)M(td|${EPOCH}) where 60 is the seconds to wait for the callee (the called party) to answer L(3600000) is the absolute limit of the call once it has been answered, in ms M(td|${EPOCH}) is the macro to execute when the call gets answered. ${EPOCH} contains the current unixtime. That's the macro: [macro-td]

I'm currently running asterisk@home v 2.7. However I believe asterisk has inbuilt a system wide speed dial system. Preserved number range starting at 300. Just wondering if it's possible to view/backup/restore/modify this data without having to enter it in manually. e.g. 300 301 12345678 (to save phone number 12345678 in speed dial 301?) I'm looking at creating a new installation

>>> Jan Stary <hans at stare.cz> schrieb am 05.11.2018 um 11:05 in Nachricht <20181105100534.GB44329 at www.stare.cz>: > (Are we off‑list now by intention?) No, just fooled by the list defaults (some need just reply, others need reply to all) > >> Did you also try to listen at the beginning, shortly before the real tone > appears in the audible spectrum?

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

One of my sites asked for a way to identify if the person they are calling on another extension is already on another call. To that end, I wrote a bit of code in the dialplan for my extensions that checks to see if the extension they are dialing has a device status that is anything other than NOT_INUSE. If the device is NOT_INUSE, then it dials the call normally. If it has a different status,

Hi , We are using Xen 4.0 version. These are the problems we are facing: 1) Centos-5 installed successfully from other installation media, Video card is detected but mouse is not working. 2) Centos-5 installed successfully from inbuilt template,but Video card is not been detected. 3) Debian 4.0 installed from other installation media, GUI is working but mouse is not working. 4) Debian 4.0

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout

I changed in asterisk.conf mindtmfduration = 50 The inter-digit duration is for this function SendDTMF when we dial out dtmf The question is, how do I change it without changing the source code? On Sat, Jun 7, 2014 at 1:00 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To

Jamie, I hear what you're saying. I don't believe this *should* be a plugin's responsibility, though it sounds like with XMMS it is. But I don't know how to fix it. Probably with enough archaeology into the XMMS source and other plugins I could find out. I'll file it in the feature requests and hope someone can get to it. Josh --- Matt Zimmerman <mdz@debian.org>

I am forwarding your request to the FLAC development mailing list. ----- Forwarded message from Jamie Heilman <jamie@audible.transient.net> ----- Date: Sun, 8 Sep 2002 16:13:32 -0700 From: Jamie Heilman <jamie@audible.transient.net> Resent-From: Jamie Heilman <jamie@audible.transient.net> To: submit@bugs.debian.org Subject: Bug#160155: gapless playback Package: xmms-flac

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

I am hereby forwarding the data & method use to calculate the Kolmogorov-Smirnov goodness of fit test made manually by me in R launguage which deffers with the actual inbuilt formula as shown below. Further I have plot the graph in R. In that graph how to add trendline (i.e. straight line passing through maximum points in plot) to a Plot. R script is as follows please run this script to see

Another two months, another winetricks release. (So much for release early, release often :-) Highlights: - fixes a bunch of verbs whose download URLs had changed, especially vcrun2005 and vcrun2008. - fixes the annoying "unexpanded variable" error which happened when running old wine with a new wineprefix. - brings back gog support (though without automated download for now; any

Hi, I've set an Asterisk 11 box with a TDM400 board and Dahdi 2.7.0.1. I've connected an FXS port to an FXO one and issued a couple of channel originate command to measure the duration Asterisk/Dahdi needs to detect a dahdi call is coming in. Basically, using EPOCH variable, I'm reading a 2 or 3s duration with the followinf AEL2 dialplan: context remote { s => { if

I had an incident, whereby the caller didn't either hang-up their SIP phone properly or the disconnect/hang-up information didn't properly find their way back to the Asterisk-PBX and it left the company phone system in intercom mode with about 90 phones overnight (624mins, CPU utilisation was running much higher than normal until i used the meetme kick <channel> all

Hi, I'm new to asterisk and have a little trouble in developing my first more complex dialplan. The basic task is a click to call solution: - call one number via sip, play some announcements, do cdr etc. and put the callee into an conference room with music on hold - call a second number via sip, play some announcements, do cdr etc. put the callee into the same conference - have a nice chat

Hi All, I've been having a hard time getting the AbsoluteTimeout function to work. Is this Function working in for SIP? I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf. I've also been running these test on ver 0.5.0 exten => _X.,1,Absolutetimeout(20) exten =>