similar to: Supporting "End User Line Features"

Displaying 20 results from an estimated 6000 matches similar to: "Supporting "End User Line Features""

2007 Sep 13
4
how to plot shaded area under a curve?
say, I am plotting x=seq(0,5,len=100) y=-(x-5)^2 plot(x,y) how can I put some color or verticle lines below the plotted curve? [[alternative HTML version deleted]]
2019 Oct 11
3
clarification on gosub, macros and AEL
I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXXXXXX.,n,Macro(fastbusy) is deprecated in favour of Gosub(). True so far? But then there are "macro"s defined in extensions.ael: macro foo() {
2010 May 18
3
About option U in Dial Ast version 1.6.2
Has any one used this? U(x[^arg[^...]]): x - Name of the subroutine to execute via Gosub arg - Arguments for the Gosub routine Execute via Gosub the routine <x> for the *called* channel before connecting to the calling channel. Arguments can be specified to the Gosub using '^' as a delimiter. The Gosub routine can set the variable ${GO
2006 Jan 28
0
Re: 5, 000 concurrent calls system rollout question
What about IAX - SIP or IAX - IAX? ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, January 28, 2006 5:43 AM Subject: Asterisk-Users Digest, Vol 18, Issue 185 > Send Asterisk-Users mailing list submissions to >
2005 Mar 01
2
Important :: Please support the development of a new Jitterbuffer for SIP
Steve Kann has developed a new jitterbuffer for IAX2, that hopefully will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable relase. Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this out and complete it in time. Please paypal your contribution to sponsor@astertest.com today. Every
2009 Mar 06
1
GoSub & Queue
I have a caller screen queue setup. Basically a caller calls in, goes through a IVR, and before that caller is put into the queue, they get a sub ran on them first asking for them to say there name. That gets saved and they are entered into the queue using Queue(mainqueue,,,,300). In the queues.conf i have a list of members these are local/extension at external-default, there are two
2009 Jun 13
1
1.6.0.10: core restart on ReceiveFax()
For our internal fax machines, I'm checking if the faxes are going to branch offices. If they are, I want to capture and email them to the branches. I've set up extension 8447 to test this. A fax machines is connected via an SPA 2102 on 173. Any calls from 173 are sent to: [outbound-fax] exten => 8447,1,Answer() exten => 8447,n,GoSub(Capture-Fax,s,1) exten
2003 Nov 07
2
Annoteting graphs using text
Dear All, I am new to R and am trying to learn how to create functions using R. Below is code which calculates Lin's Concordance Coefficient. After I calculate the coefficient I want to create a scatter plot which annotates the coefficient along with preceding text onto the plot. The below code doesn't seem to work. If I use only the object 'lincc' on the text command it works
2011 Apr 03
1
From 1.4 to 1.8: stdexten issue
Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten =>
2023 Jun 17
1
Expanding my answering-machine system
On 6/17/23 08:47, Steve Matzura wrote: > > Both Background() and WaitExten()  allow the caller to enter DTMF > digits. Asterisk then attempts to find an extension in the current > context that matches the digits that the caller entered. If Asterisk > finds a match, it will send the call to that extension. > > > My question then is, is "*" a valid exension, as
2010 Jul 23
2
application call to Gosub affects flow of control, and needs to be re-written using AEL
Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "ael reload" command trigger a lot of warning like that "application call
2013 Aug 08
1
queue member ackcall - cpuspikes
hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the caller. Language: AEL Similar Example:http://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall Scenario: 1. User calls in a General Number
2012 Oct 31
1
Asterisk 11 and stdexten written in AEL invoked by pbx_config
Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten
2009 Feb 24
3
Gosub behavior change <=1.6.0.5 to 1.6.0.6
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Here's one that may be of interest to any upgraders. If you rely on the behavior of gosub you may want to make note of this change. I have an incoming call context: exten => _XXXX,n,GoSub(incoming,${EXTEN},1(${EXTEN})); that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the
2023 Jun 17
1
Expanding my answering-machine system
OK, this is how I thought it's supposed to work. It just confounded me why the book would say the Playback() and Background() syntax were the same, then in the very next paragraph give an example that belied that claim. On 6/17/2023 1:46 PM, Doug Lytle wrote: > On 6/17/23 08:47, Steve Matzura wrote: >> >> Both Background()  and WaitExten()  allow the caller to enter DTMF
2016 Mar 03
2
Asterisk Call Forwarding
Hi, Thanks Phil, I will implement this and get back to you. Best Regards, Madushan On Thu, Mar 3, 2016 at 4:12 PM, Phil Reynolds < phil-asterisk at tinsleyviaduct.com> wrote: > On Thu, 3 Mar 2016 08:21:14 +0530 > Madushan Geethanga <mgliyanage.rc at gmail.com> wrote: > > > Hi > > I have to setup call forwarding. How do we setup Call forwarding in > >
2011 Aug 15
3
Queue Breakout Input being Ignored
Hello, Raw stats: Version:1.8.3.2 OS:Centos 5.6 Special setup: postgre database I am having a few queue issues with Asterisk specifically relating to breaking out from queues while on hold. The intent is that while someone is on hold they can press a key (lets say *) to break from the queue and go elsewhere (in this case to leave a message). However In all of my testing I am unable to get
2009 May 08
0
Can't GOSUB_RESULT with Dial U() option ...
Hello, I'm not understanding how to use GOSUB_RESULT in U() option from Dial app (I'm using 1.6.1) My extensions.ael is : context mylocal { 2 => { Dial(SIP/7530,,U(mynotify)); HangUp(); }; 3 => { Dial(SIP/7531); HangUp(); }; }; macro mynotify () { GOSUB_RESULT=ABORT; }; I
2008 Feb 29
0
1EZphone is only two way browser softphone - SIP Softphones and Citrix ?
Yes, try http://1ezphone.com its a browser softphone. ----- Original Message ----- From: Zoa To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] SIP Softphones and Citrix ? Date: Fri, 01 Feb 2008 23:09:56 +0200 I'm working for zoiper.com and i'm willing to help out with ours when needed. Zoa d4rk f1br wrote: >
2013 Mar 12
0
Calls getting "stuck open"
I have a system running Asterisk 11.2.1 that has had a couple calls between internal extensions get "stuck open". I didn't catch the verbose log for the first one, since I generally don't verbosely log to file, but the second one shows that the call that got stuck was dialed, but the caller hung up before the called device answered. This server is running a hotdesking