Displaying 20 results from an estimated 1000 matches similar to: "Problem in Recording file on Hangup ?????"
2007 Dec 18
1
Call Recording on Hanup
Hello everyone out there, I am having a problem in call recording with php
agi library. I have already recorded voice after playing an IVR, to accept
the recording user need to press one. but I need to record a call on hangup,
Is there any way to do it. Currently i am using record_file() function in
php. Is there any way to record voice by using record_file() function with
hangup. can anyone helps
2007 Oct 11
0
Problem in placing Call with Asterisk (Got SIP response 500 "Internal Server Error")
Hi guys this is my Ist mail on this group, I am running asterisk with CentOS
4.4 machine. When i initiate a call then error message apears. calling
Number is provided to Asterisk by the php application. Error message appears
like this
Got SIP response 500 "Internal Server Error" back from 209.47.92.xxxx
[Oct 10 23:55:20] WARNING[19289]: pbx.c:4976 ast_pbx_outgoing_exten:
2007 Mar 09
2
Is there any variable for Voicemail Password in Asterisk
Hi guys
This is my Ist post on this group. Is there any variable like ($VM_CALLERID
for voicemail mailbox) for accessing Asterisk Voicemail password which is
set through comedian mail.??????????????
plz reply me as soon as possible....
<html><div><PRE class=quote><IMG height=2
src="http://graphics.hotmail.com/greypixel.gif" width="100%"
2007 Oct 31
0
Jamshed Zaidi wants to chat
-----------------------------------------------------------------------
Jamshed Zaidi wants to stay in better touch using some of Google's coolest new
products.
If you already have Gmail or Google Talk, visit:
http://mail.google.com/mail/b-41f7263365-9f176e071a-d710297ddb46eb39
You'll need to click this link to be able to chat with Jamshed Zaidi.
To get Gmail - a free email account from
2007 May 16
0
AGI "record_file" issue... bug?
I am having a problem with "record_file" working properly depending on when
it is called -- basically if it is called immediately upon a call, it acts
like it does not hear anything from the callers phone (yes, my phone is
setup properly and functions fine otherwise)... if I do a "background" or
"festival" command before calling it, it works fine.
Details below:
2014 Nov 14
0
Asterisk 13 confbridge recordings not working
We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13.
Here is the dialplan segment
same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes))
same =>
2011 Apr 12
0
Help required
Hi Sadaf,
Out of curiosity, what sorts of things have you tried to fix this?
For example, after playing around with this a bit, if I remove your
"eps" parameter from your `ranges` list, it works.
Perhaps you should try tweaking the values you pick for your
parameters. You don't even have to put it in the `tune` function to
get an idea of the ranges you should use:
R>
2010 Nov 23
1
wideband recording in Asterisk 1.8
Hi all,
I would like to know if something I am trying to do is possible.
I currently using 1.8 and would like to make recordings in wideband
(16khz)
I have an aastra 6739i which supports the g722 codec.
I made an agi application in which I would like an user to record a
promptlist.
I use the agi command record_file.
Which format setting do I need to be able to record in wideband?
2023 Apr 02
1
Count matrix of GSE146049
How can I subscribe to R genomic list?
On Sun, 2 Apr 2023, 9:28 pm Peter Langfelder, <peter.langfelder at gmail.com>
wrote:
> It's a microarray data set, so I don't think you would want to apply
> an RNA-seq pipeline. You'd be better off applying a normalization
> appropriate for this type of microarray data.
>
> HTH,
>
> Peter
>
> On Sun, Apr 2, 2023
2011 Apr 04
1
Problem using svm.tune
Dear Sir,
I am stuck with a nagging problem in using R for SVM regression. My data has 5
dimensions and 400 observations. The independent variables are :
Peb, Ksub, Sub, and Xtt.
The dependent variable is: Rexp.
I tried using the svm.tune function as well as <_tune(svm.....), to tune the
hyper parameters: gamma, epsilon and C.
Since I am new to R, I am probably not using the svm.tune
2006 Feb 08
1
Possible AGI Bug in Asterisk?
Dear All,
I seem to have stumbled across an AGI problem;
I have written an AGI Script (bottom of this email);
The script does the following;
Makes a CDR entry when called
Records the call
Updates the CDR
Finds a corresponding DNIS from the SMDR table (captured via a serial
port logger)
Matches up the record and updates the CDR.
The script works perfectly in my test lab and has been doing so
2011 Apr 04
2
Please help
Dear Sir/Madam,
I am stuck with a nagging problem in using R for SVM regression. My data has 5
dimensions and 400 observations. The independent variables are :
Peb, Ksub, Sub, and Xtt.
The dependent variable is: Rexp.
I tried using the svm.tune function to tune the hyper parameters: gamma, epsilon and C.
I am getting the following error message:
Error in predict.svm(ret, xhold,
2023 Apr 02
1
Count matrix of GSE146049
It's a microarray data set, so I don't think you would want to apply
an RNA-seq pipeline. You'd be better off applying a normalization
appropriate for this type of microarray data.
HTH,
Peter
On Sun, Apr 2, 2023 at 11:09?PM Anas Jamshed <anasjamshed1994 at gmail.com> wrote:
>
> I want to get the count matrix of genes from
>
2008 Dec 16
1
Callback when objects connect as a habtm relationship
If Product and Category models are in a habtm relationship, i.e.
class Product < ActiveRecord::Base
has_and_belongs_to_many :categories
end
class Category < ActiveRecord::Base
has_and_belongs_to_many :products
end
I want a piece of code to be executed every time a product is connected to a
category.
Where do I put this code? Which callback (and on which model) will be
triggered?
Any
2011 Nov 04
1
problem when exiting from "record file" function without pressing the escape digit
Hi everybody,
I've been working on a project which records the voice of the incoming call.
I use record_file function of asterisk as described below:
RECORD FILE <filename> <format> <escape digits> <timeout> [offset samples]
[BEEP] [s=<silence>]
filename: record1
format: wav
escape digits: #
timeout: -1
offset samples: 0
BEEP: 1
silence: 3000
*Please read the
2006 Mar 10
0
Background timeout and Read questions
Hello list,
Three questions on dialplan commands/functions:
- cmd BACKGROUND:
a) If i just want to allow user to send DTMF ONLY while the message is
played (with no additional time after ), does TIMEOUT RESPONSE = 0 make
sense or this action produce some border effect ?
b) if YES (border effect = stupid me), may i use an AGI with
'record_file' and 'get_data' to do something
2006 Nov 02
1
AGI Problems
Hi,
I've got a setup whereby calls come into the asterisk server (1.2.7.1)
over a IAX2 trunk and into a dialplan that launches a php AGI script:
[live-full]
exten => _X.,1,Set(TIMEOUT(absolute)=0)
exten => _X.,2,NoOp(${EXTEN})
exten => _X.,3,DEADAGI(live-full.php)
exten => _X.,4,Wait,2
exten => _X.,5,Hangup
The script is using phpagi-2 from http://phpagi.sourceforge.net/ and
2023 May 26
1
Function DENOISE not registered
Hello,
when I call my conference, I see this error in my logs:
ERROR: Function DENOISE not registered
here is snippet from extensions.conf
...
same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes)
same => n,Set(CONFBRIDGE(bridge,record_conference)=yes)
same => n,Set(CONFBRIDGE(bridge,record_file)=/home/asterisk/file.wav)
same => n,ConfBridge(1000)
same =>
2004 Jul 16
0
Samba Read Only
i used samba 3.0.2 on RedHat 9 ( my Server )
my client using JAMD ( kernel 2.4 ) oke no problem, to mount share file we
using LinNeighborhood
and another my client using LormaLinux ( kernel 2.6.7 ), can mount the
share file, but went i open
file ( using open openoffice ) that file just read only
any solutions
regard's
zaidi
2012 Oct 12
0
[LLVMdev] Cambridge LLVM Afternoon Workshop on Monday
Hello the lists,
Please feel free to ignore this message if you are not in Cambridge.
As previously mentioned, we are hosting an afternoon LLVM workshop in the Computer Lab on Monday (15th). We'll start in the morning with a tutorial and then have 15-minute talks with 5 minutes for questions and speaker changeover in the afternoon.
The event is open to all and will be in room FW11 of the