Displaying 20 results from an estimated 1000 matches similar to: "Tutorial: compiling and installing Asterisk 13"
2013 Dec 30
0
Couple of new tutorials on asterisk 12 and ARI
Hi all,
I put together a couple of new tutorials on compiling Asterisk 12 with
PJSIP on CentOS 6.5 and test-driving ARI on the same box.
You can find them at:
http://astrecipes.net/index.php?q=AstRecipes/Compiling%20Asterisk%2012%20on%20CentOS%206.5
and
http://astrecipes.net/index.php?q=AstRecipes/Getting%20started%20with%20ARI
Comments welcome and happy holidays! :)
l.
--
Loway
2009 May 25
1
New tutorial: storing audio recordings per day
Hi everyone,
after doing the same thing multiple times and struggling to remember how it
was done, I have prepared a small tutorial that explains how to save
monitored files in different folders per day. This is quite useful
becausethe resultingfile system is way more manageable than having maybe
100,000 files all saved in the same folder.
You can find the tutorial here:
2015 Apr 23
2
Sample Docker images for Asterisk available
Hello all,
I created a set of Docker images running Asterisk and exposing AMI /
ARI ports that i found to be quite useful for ARI / AMI development
and regression.
As they are based on Docker with whaleware, adding new configuration
files to roll your own dialplan / queues / voicemail etc is pretty
easy. And you can run quite a lot on the same box to simulate
clusters.
There is no SIP / RTP
2013 May 14
4
dial and bridge
Hi all,
I need some advice - I have been working on originating multiple calls
using AMI and then joining them.
What I want to do is:
- dial call 1 (where the caller is in a "channel" format, like SIp/1234 or
Local/1234 at ext) and "park" it somehow
- dial call 2 (where again the caller is in channel format) and join it to
the previous call.
As a requirement, I cannot use the
2013 May 13
1
amiDebugger - might make your life easier if you program through the AMI
Hi all,
I have been playing with the AMI quite a bit lately - mostly debugging
WombatDialer in production, but that's a different story - and I have been
frustrated by the lack of a simple way to interact CLI-like with the AMI
itself. So I have decided to write something myself to make my life easier,
or at least a bit less miserable.
The result is a little webapp that you can use as a sort
2016 Jun 14
4
Pet project: one step Asterisk compile on Centos 7
Hi all,
I thought I'd share I script I made (based on some of Leif's works)
that lets you download, compile and install Asterisk all in one go;
and then removed the dev tools used.
We use it quite a bit to provision systems using Ansible, but it is
easier than remembering everything every time even if you are using a
shell.
At the moment I have scripts for Centos 7 and Asterisk 13, but
2013 May 31
1
WebRTC softphone for Asterisk - any suggestion?
Hi All,
I wonder if any of you has some suggestions on which WebRTC
client/softphone to use for a click-to-dial, webpage hosted solution. Any
suggestions?
Thanks
l.
--
Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com
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2017 Feb 12
2
compiling asterisk-14.3.0-rc2
hi all,
can someone help? I have centos 6.8 trying to install asterisk 14.3.0-rc2
on it with options as stated below -
./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib
--with-jansson=/ --with-pjproject-bundled
when I tried to run "make menuselect". i get the error below.
Makefile:109: makeopts: No such file or directory
****
**** The configure script must be executed
2006 Nov 09
2
A couple of new tutorials: installing * 1.4 and the Asterisk GUI
Hello list,
I have prepared a couple of new tutorials you may find interesting:
- Installing an Asterisk 1.4 beta system - at http://astrecipes.net/?n=216
- Installing the Digium's Asterisk GUI for 1.4 - at
http://astrecipes.net/?n=217
It's nothing too complex, but you may find them interesting, especially
the new Asterisk GUI.
Any comment is welcome - the site is a wiki, so feel
2007 Feb 22
3
New tutorial: DTMF tone detection
Hello list,
I have prepared a small tutorial today that deals with how to avoid
Asterisk rebuilding DTMF tones when using it to connect industial
appliances that use DTMF. You can find it at:
http://astrecipes.net/index.php?n=248
I know it isn't everybody's piece of cake, but I thought somebody could be
interested as well :)
l.
--
Home of QueueMetrics -
2007 Aug 09
1
a couple of new tutorials
Hello list,
I posted a couple of tutorials lately, maybe someone can benefit from them:
The first tutorial explains how to transform your Asterisk call recordings
(in WAV or GSM) to lo-fi MP3 to save a lot of space. It's actually pretty
easy to implement using a makefile.
http://astrecipes.net/index.php?n=294
The other tutorial lets you implement a way to monitor all outgoing
traffic
2005 Oct 17
4
compiling Asterisk 1.2 with zaptel and h.323
Hello list,
I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with
a TDM400 card and H.323.
You can find it at http://www.oinko.net/astrecipes/index.php?n=102
Any comment / suggestion / modification /bugfix is welcome!
I was wondering: is there any way to build a version of Bristuff for 1.2
beta 1?
Bye for now,
l.
--
Loway Research - Home of QueueMetrics
2008 Mar 28
2
New Tutorial: Asterisk on EPIA VIA C3
Hello list,
after spending the best part of an afternoon trying to build Asterisk on
an old EPIA VIA C3, I thought that writing a tutorial would make life
easier for future compilers:
http://astrecipes.net/index.php?n=356
I had never compiled Asterisk for a different architecture, and I'm pretty
disappointed at how complex it is - building Zaptel, Libpri and Asterisk
requires
2006 Mar 13
1
music on hold without mpg123
Hello list,
after the last time that mpg123 wen ballistic on our production system, we
decided to skip mp3 playback altogether and to go for raw files. After
half an hour playing with mpg123 and sox parameters in order to translate
a mp3 file to a wav file that can be streamed back through * with no need
for an mp3 decoder, I thought I'd post the result to the list to avoid
wasting
2013 Sep 30
0
QueueWiz - a free call-center simulator tool for Asterisk
Hello all,
next week it's Astricon 10 time, so we thought we'd create something
that the community could like and use for free. It's a pretty
effective tool if you run a call-center or plan to run one.
QueueWiz is the first free web app for interactive, quick and accurate
call center sizing, cost and revenue simulation. Insert your data with
the intuitive interface, measure traffic
2006 Apr 12
1
Recording queue transfers
Regarding this article (1) I have one question to make. What can I do to
record the call if the agent makes a transfer using the "flash" button
instead of "transfer button" or using blindxfer or atxfer defined in
features. conf
If the agent makes the transfer with "flash", the comunication between the
person who is calling and is already in the queue and the target
2017 Feb 12
2
compiling asterisk-14.3.0-rc2
Thanks.
The configure run successfully.
but I got the warning below..
checking for the ability of -lsrtp to be linked in a shared object... no
configure: WARNING: ***
configure: WARNING: *** libsrtp could not be linked as a shared object.
configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp
configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr
configure:
2013 Sep 20
0
Astricon - let's talk call centers?
Hi list,
I know it's a bit OT, but for those who will be at the Astricon, we
are organizing a very informal meeting (maybe in front of a pint or
two) to talk about Asterisk for call-centers. No marketing or anything
- just a way to exchange ideas and meet f2f.
I created a facebook group to organize it - see
https://www.facebook.com/groups/507826572618269/
See you in Atlanta!
l.
--
2009 Feb 18
2
Setting SIP header on agent calls made by a queue
Hello list,
I am trying to set a custom SIP header on all calls that are made by the app
queue because I want to track a certain state at the SIP level.
If I use the following code:
exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten => s,n,Queue(myQueue)
this works fine for the FIRST call made from the queue to an agent; but if
that call does not go through, it's not repeated
2015 Jun 03
1
sslv3 alert unexpected message
hello,
my webrtc calls ends after ~60seconds with "res_rtp_asterisk.c: DTLS
failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert
unexpected message', terminating". any ideas where can be problem? or
howto debug this problem?
asterisk13.4.0-rc1 + sipml5 latest (chrome,firefox)
--
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Marek Cervenka