Displaying 20 results from an estimated 20 matches for "ruddi".
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rudi
2013 Dec 04
5
Asterisk SIP server on windows
Hi all,
I need to build an application that will be an SIP server program that will
run on Linux and Windows.
The sip server need only some features such as be able to :
- Register sip endpoints
- Answer a call and play a local file
- Make a dial from one channel to another.
I know asterisk can be stripped to exactly fit my needs. I would like to
know if there
2003 Dec 11
1
(no subject)
Hola
Hace poco comence a utilizar R y tengo dudas como utilizar el paquete de tree
saludos ruddi
---------------------------------
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[[alternative HTML version deleted]]
2001 Mar 22
1
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2007 Aug 11
1
Connecting to database on statup
Hello,
Q/ Is it possible to create a DBMS connection automatically on startup of R? (Making sure of course that the db server has been started...)
I am running MySQL on Mac OS X 10.4.2 with R2.4.1.
I have tried to write a function using the RMySQL commands (below) and place them in .First of .RProfile:
drv <- dbDriver("MySQL")
dbcon <- dbConnect(drv, {other parameters present in
2013 Dec 04
2
Unmute all users in Meetme conference as admin
Hi,
I setup an MeetMe conference.
So, the admin user calls and enter the conference in talk/listen mode.
(Options : dAaxs)
Then other users call the same conference and enters in muted mode
(options: dlmx)
How can the admin user decide, when he is ready to let everybody speaks ?
I didn't find such option in the admin menu.
Thanks
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An HTML
2012 Aug 10
3
Vector size limit for table() in R-2.15.1
Hi,
First, thanks in advance. Some useful info:
>version
platform x86_64-unknown-linux-gnu
arch x86_64
os linux-gnu
system x86_64, linux-gnu
version.string R version 2.15.1 (2012-06-22)
I'm trying to use the table() function on a 2 column matrix that has 711
million rows (see below). However, it freezes. If I subset the matrix to be
less than or equal
2008 Nov 03
0
asterisk src=dst
Hi all
I saw in the CDR stocked in mysql as well as those in the csv file that
some time, the src field is the same as the dst field which is the
extension.
When does it happens.
Here, we have 4 dgits extensions and most of the time the dst field is
the extension and the src field is the 10 digit customer phone number.
Do you know when does this happens ??
Thanks
Ruddy Gbaguidi
2011 Jun 11
1
Full SIP dial string
Hi All
I want to be able to read some sip informations (from a database) like
username, password, host and extension number and place a Dial from
asterisk.
So basicly, I want to dial sip extensions without modifying sip.conf each
time.
I don't know, in the dialplan, what the dial string should look like.
I tried
SIP/<username>:<password>@<host>/<exten>
2014 Feb 21
1
Cancel a ringing SIP call when the other party disconnect
Hi,
Here is my scenario.
I have a SIP call between two SIP endpoints. A calls B.
During the ringing, B disconnects (network cable is unplugged).
But A continue ringing forever (until the dial timeout) even if asterisk
detects that B is disconnected with the qualify.
Is there any setup or asterisk configuration I need to enable to have A
close its call ?
Note: when A is already talking with B,
2019 Oct 08
0
Asterisk 13.29.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 13.29.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.29.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
2019 Oct 08
0
Asterisk 16.6.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
2019 Dec 23
0
Asterisk 17.1.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 17.1.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
2012 Oct 25
6
How to tie orders taken to specific CDR records
Our phone operators work off of an Asterisk queue. They take calls from
customers and take orders with our back end systems. What I need to be
able to do is tie the orders taken to the specific CDR record that
reflects the call from which the order originated.
The typical/sample CDR table doesn't have a primary key. I can add an
auto-generated PK, but the CDR is not written until the
2008 Aug 21
3
After Dial execution, using DIALEDTIME, ANSWEREDTIME
Hi,
I noticed that when dial terminates it does not return to the dialplan,
and therefore can not execute any entry after Dial().
Is there any trick to overcome this limitation ?
How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if
I can not execute anything after Dial()?
I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls
end
2009 Apr 23
9
AMD Not Working
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
can any one suggest us, what might be the problem
and possible solution to it.
below is the log
-- Executing AMD("SIP/sip-ffe0", "") in new stack
-- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4)
Apr 23 08:00:26
2020 Apr 30
0
Certified Asterisk 16.8-cert1 Now Available
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following
2020 Apr 30
0
Certified Asterisk 16.8-cert1 Now Available
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following
2009 Jan 16
0
No subject
AGI is executable.
=20
Then run 'agi debug' from the asterisk cli, place a call and see what
was send and receive from your agi
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A.
Shigley
Sent: April-23-09 12:26 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] AGI PHP script
=20
I have the
2020 Oct 20
2
Asterisk 18.0.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.0.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
2008 Jul 31
0
Asterisk CDR "**Unknow**" as channel name
Hi all
I have been looking at my asterisk CDR in the mysql database and
some channel names are set to "**Unknown**" string.
When I look at the code, everybody when calling ast_channel_alloc set a
channel format
like SIP/%s or Zap/%s
Only app_voicemail.c doesn't when sending emails and I don't use voicemail.
Why app_voicemail needs to allocate a channel to send emails ?
And in