Displaying 20 results from an estimated 10000 matches similar to: "Continue script on hangup"
2020 Jan 24
4
Perl AGI: read variable with quotes
Hi Gang
I have stumbled of this problem.
I need the P-Asserted-Identity header in an AGI scrip.
In the Dial-Plan I do:
same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)})
In the AGI I do:
my $pai = $AGI->get_variable(PAI);
This works fine, unless the PAI contains quotes:
P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone>
I get "<sip:1000 at
2009 Jan 27
2
RFC -- Improving the quality of the mailing lists
The -user and -dev mailing lists are a valuable resource -- when they are
not cluttered by posts unrelated to the "charter" of the lists.
In my limited memory, this last weekend represents a new low in the
"relevant subject to noise ratio."
Replying to requests with meaningless, misleading, or misspelled subject
lines ("I need help," "asterisk help,"
2010 Jul 08
3
Not detecting hangup
We have had 20 calls over the last month where the SIP channel has not
identified that the person on the receiving end has hung up.
Is there a way of fixing this ?
TIA
Julian
2010 Nov 27
3
How to hangup all channels
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
I want to use the teleyapper system for broadcasting call for security reason but i need that all channels are free when a security call is ready to start!
I already search in the old post without success.
Can anyone help me?
Thanks and sorry for my newbie english
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2014 Aug 22
1
Can't hangup channel from CLI
Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting
Asterisk from a Tekelec T9000.
I'm accumulating stuck channels.
I'm googling now and I recognize that Friday afternoons are the worst time
to ask questions, but I'm getting desperate because this is keeping me
from rolling a system out to production. (Yup, I know. Who rolls out a
system on a Friday
2007 Dec 06
3
CDR Function in Hangup Channel
So... I'm trying to access CDR(duration) and CDR(billsec) inside h...
I keep getting 0. Can I access the CDR function inside a hangup extensions?
Asterisk 1.4.13
Thanks, Doug.
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2016 Apr 13
5
recreating extensions.conf from live dialplan ?
with the slip of a finger, i destroyed by extensions.conf (grep -i >
extensions.conf)
I have a backup that is dozens of hours of code old.
is there a way i can use the asterisk cli (or some other asterisky
method) to recreate that extensions.conf ?
2007 Apr 09
3
Play audio and continue to next priority before audio ends...
Hello list members.
I would like to know how to playback an audio file to the caller, and while
it's played asterisk to continue executing the next priorities on
extensions.conf
That's not the case when using "playback" command, because the next priority
is executed until the audio file ends playing. I want to evaluate some
variables while caller hears the audio file.
Any
2010 Jul 04
1
Anyway to know when a channel is going to hangup if Dial Timeout option is used?
Hi Guys,
I have a channel that is dialed with *Timeout* option. So, there is definite
time to it. Only thing is that I don't have control of that channel. I only
know that it's using g729 codec and that there is only one channel that is
using g729 at any given time. So, my question is:
2006 Oct 13
3
error running webserver 7 with the DTrace dvm agents...
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
<title></title>
</head>
<body text="#330000" bgcolor="#ffffff">
<tt><font size="+1">I am attempting to run the sun webserver 7
2010 Oct 25
2
Re : thousands Hangup per second /saturation of bandwidth
Any news for this problem.
Who has this problem
Why you don't answer.
--- En date de?: Jeu 21.10.10, ALAEDDINE abbech <alasupcom at yahoo.fr> a ?crit?:
De: ALAEDDINE abbech <alasupcom at yahoo.fr>
Objet: thousands Hangup per second /saturation of bandwidth
?: asterisk-users at lists.digium.com
Date: Jeudi 21 octobre 2010, 11h42
Hello,
I have a problem of saturation of
2014 May 05
2
how to hangup Local/100 channel
Hello All,
one of the extensions fall into a loop, I don't know how to hangup that
channel
-- Executing [i at autoatten:2] Goto("Local/100 at sipphones-000001b2;2",
"s,2") in new stack
-- Goto (autoatten,s,2)
-- Sent into invalid extension 's' in context 'autoatten' on
Local/200 at sipphones-000001b2;2
-- Executing [i at autoatten:1]
2009 Apr 20
2
Execute after hangup
What is the syntax to progress with a dial plan after hangup please?
Michael
2011 May 17
5
Skype-like dialing from web page
Hi,
Is there any softphone or TAPI plug-in that allows one to dial from a web
page? As you may know, Skype has a mechanism that converts phone numbers on
a web page to a click-to-dial application. I'd like to use this but on a
normal softphone (Bria, Xlite, other).
Regards,
Mike
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2016 Apr 13
4
recreating extensions.conf from live dialplan ?
On 4/13/16 11:57 AM, A J Stiles wrote:
> You could try
> *CLI> dialplan show
Between my older backup and dialplan show, I guess that's my best shot.
Thanks :D
2017 Feb 06
3
Call List Campaign to an IVR
Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg.
If you wanted you could leave it ringing for twenty minutes and it would still have the same effect.
Kind regards,
Matt
> On Feb 6, 2017, at 12:29 PM, Tech Support <asterisk at voipbusiness.us> wrote:
>
2014 Nov 27
3
day night service toggle
Hi,
I need dialplan to set INCOMING call forwarding during lunch break to my secretary.
I want that I can set call forwarding by dialing an extension number to turn it ON or OFF.
I am using asterisk 11.
Thanks
Abdullah Faheem
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2017 Feb 06
3
Call List Campaign to an IVR
> On Mon, 6 Feb 2017, Tech Support wrote:
>
> We were able to develop a feature to send the call to voicemail about 90% of the time. That way, an end user could (1) not be bothered by having to answer the call, (2)
> delete the message without listening to it, or (3) listen to the message when it was most convenient for them. That way, they were in control and things were
2017 May 10
7
How to detect fake CallerID? (8xx?)
I have a 'time and attendance' application. Think janitorial or security
kind of thing where an employee goes from location to location.
They're supposed to 'clock in' when they get to a site using a phone at
that site to prove they're there.
Some employees have discovered 'fake caller ID' services can be used to
say they're on site when they are not.
How
2012 Oct 18
2
Problems with AGI and existing channel
Hi,
"Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2012-03-08 23:05:09 UTC"
We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a "hickup" and then continues.
We are using AGI to lookup incoming calls in directory.
It is kinda annoying, and I don't understand how it can be