Displaying 20 results from an estimated 20000 matches similar to: "Call duration limits not working"
2011 Jan 17
1
Max call duration
I've searched through the wiki but I can't find what I need...I'm trying to figure out what the max call duation is. I found references to "show application AbsoluteTimeout" but that isn't in 1.6 (not even prepending "core" to the front). A core help show didn't help...
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2007 Mar 21
2
Limit call duration
Hi everyone,
I'm new to Asterisk, but I like it ;o)
Have a question to you;
How can I limit the incoming call duration?
--
Suich
2009 Jun 10
0
Dial option limit call duration
Hi,
we're using the limit option like this:
Dial ....L(60000:30000)
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] --
Limit Data for this call:
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] >
timelimit = 60000
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] >
play_warning = 30000
[Jun 10 16:14:41] VERBOSE[12196]
2005 Sep 19
0
Dial time limit doesn't work when calling party transfers
Hi,
I'm using the AbsoluteTimeout and Dial (with L() option) commands to set
a timeout for my calls, but when the calling user transfers a call the
timeout doesn't work and the call last until hanging-up.
If the call is not transfered the limit works just fine.
?How can I make this work?
Thanks in advance.
My asterisk version is 1-0-9-07 and here's an example of one of the
macros
2006 Jun 26
1
M() option to Dial
I'm using the M() option to Dial() and having problems. In the
following dialplan example ANY digit exits the macro. When the callee
presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run. Does
anyone have any ideas as to what I'm doing wrong? Asterisk 1.2.x
[extensions]
exten => 2998,1,Dial(Zap/1/5551212,,wM(answer-confirmation^20))
[macro-answer-confirmation]
exten
2005 Mar 23
1
Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up.
Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k is installed
I have a x100p card and it doesn't detect a hangup from the calling
party when going in voicemail(). My PSTN provider is sending open loop
disconnect (voltage decrease for a given moment of time). Actually
Progress Detection is HIGHLY EXPERIMENTAL so it should not be required
to fix this problem. I wonder if disconnect supervision is the
2003 Oct 26
0
Re: [Asterisk-Dev] important feature missing?!
>one of the common features in PBX devices is the ability to set a time limit
>on duration of calls (esp. outgoing, for each station) and usually a warning
>beep is played few seconds before time runs out. as far as i could
>understand it's possible to set a time limit on calls using something like:
>exten => 2000,1,AbsoluteTimeout(20)
>but that's only on *incoming*
2005 Mar 23
0
Re: [0] Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up.
Rich Adamson <radamson@routers.com> wrote on 2005-03-23 09:08:
>>
>> I have a x100p card and it doesn't detect a hangup from the calling
>> party when going in voicemail(). My PSTN provider is sending open
>> loop disconnect (voltage decrease for a given moment of time).
>> Actually Progress Detection is HIGHLY EXPERIMENTAL so it should not
>>
2007 Jan 17
2
AbsoluteTimeout with canreinvite=yes
Is AbsoluteTimeout designed to work with canreinvite=yes?
If not, are the any other options for disconnecting a call after a
predefined duration when using canreinvite=yes?
Thanks!
David
2007 Feb 22
1
GotoIf DURATION
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Helvetica, Arial, sans-serif">Hi,<br>
<br>
I am
2005 Oct 10
0
Asterisk behaving wierd!!
hello,
I have been using asterisk now for about 2 years now on a RH8.0 it is our
main call gateway.
I have on the box 3 T1 TDM cards connected to 2 Rhino channel
banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA
186s.
It has been working good till today some few hours ago. i just
discovered that there were no dialtone on the phones.
Asterisk did not spit out any error, it
2008 Aug 15
1
Vectorization of duration of the game in the gambler ruin's problem
Hey fellas:
In the context of the gambler's ruin problem, the following R code obtains the mean duration of the game, in turns:
# total.capital is a constant, an arbitrary positive integer
# initial.capital is a constant, an arbitrary positive integer between, and not including
# 0 and total.capital
# p is the probability of winning 1$ on each turn
# 1-p is the probability of loosing 1$
# N
2003 Dec 18
1
FreeBSD mknod errors
Hi all, I ran across a few threads on google regarding FreeBSD's rsync mknod
problem but I didn't find any good fix. I was wondering if anyone knows more
about this and a good fix. Here's an example of the output.
mknod qmail/queue/lock/trigger : Invalid argument
mknod qmail/supervise/qmail-send/log/supervise/control : Invalid argument
mknod
2006 Mar 07
0
a2billing problem with call duration
Regards!
During the use of areski a2billing software I'm getting same problem all the time.
Actually, after 15 minutes of speaking to someone over calling card, connection brakes.
Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication.
In the logs everything seems to be fine.
I'am sending You
2004 Jul 15
1
"Reverse Hold" feature prototype...
I have no idea what this really should be called, so for lack of a
better name, I called it "reverse hold". Hopefully someone else can make
use of it, or even make it better, as its the first thing of its kind
I've made for asterisk.
Like most people, I'm very busy, so when I call other companies, sitting
on hold really sucks. If you have speaker phone, its not so bad, but
then
2007 Apr 16
3
duration sec and billing sec in cdr
Hi guys,
i've installed asterisk to handle multiple voip accounts. I've looked
at CDR configs, and managed to have cdr-csv files growing after each
call. It would be easier to check my locak asterisk cdr's than logging
into each account and check them at the provider website.
i found that if i ring my sip softphone from my ata, bill seconds are
counted correctly. however, if i
2011 Jan 29
1
Can a duration limit be specified in spool call file?
Hi Everyone,
I don't see any parameter for limiting duration of a call in the .call file
for Asterisk spool outgoing directory.
I'd rather not use a MeetMe to drop the call in a conference room and to
then limit the call duration as that complicates things unnecessarily.
I am wondering if there is anything else I can do or if the "Channel"
parameter take call duration like the
2014 Jan 08
2
Call duration limit ? Calls end after 15 minutes...
Hello,
I see the strange behaviour that outgoing calls end after 15 minutes.
I didn't knew there is some kind of call duration limit that can be set ?
Is there ?
Using Asterisk 1.8.12.2
Kind regards,
Jonas.
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2002 Jun 02
0
daemontools supervise and smbd
Hello,
I'm trying to run smbd with djb daemontools' supervise.
Here's the way I followed:
# 2/6/2002
# problem running samba-2.2.4 and supervise
problem: run smbd with supervise
subproblem 1: smbd & nnmbd with -D option deteach from tty, so
supervise cannot control them;
solution 1: we can patch smb/server.c (foreground patch) to prevent
deteach, but a second
2005 Jan 03
0
Limit max calls & call duration
Hello,
I was wondering if there is a simple way to limit the number of simultaneous
calls in an Asterisk PBX ?
I've seen that we can make this easily per channel (like in SIP.CONF) :
incominglimit=X, but I'm looking to limit the maximum calls all channels
together.
Another thing. Working with asterisk-perl, I need to get the call
duration, currently
I use