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Astertest - asterisk stress testing tool

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9. Astertest - asterisk stress testing tool


Astertest is a Windows application that can test the CPU load of your Asterisk PBX server. In
order to use it you must have advanced knowledge in VoIP. To complete the test you must
have an Asterisk PBX server that originates the calls and one more Asterisk server which to
be tested. You can download the application from http://astertest.com/downloads/. You will
need most of the things there. Let us start doing the things one by one.

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1.Select the servers


You must have clear idea which machine will be used as Origination server and which one
as Test server.
2.Audio Files
The audio files are stored in /usr/src/asterisk/sounds/. Copy astertest-rtp.gsm from
http://astertest.com/downloads/ to the Origination server. On the Test server you have to
copy all the sound file formats you wish to test. For alaw you need some *.alaw audio file for
example - ast-test-al.al. You can use your own files with different name, but make sure
when you configure the extensions.conf on the Test server to use the correct audio file
name with the Playback command, ast-test-gsm.gsm for gsm tests, ast-test-ul.ul for ulaw,
ast-test-ilbc.ilbc etc. Now when you have the audio files for the test you have to implement
some new modules in asterisk that will give you the information for the actual CPU load
while testing.
3.New Asterisk Modules
The modules below must be installed on both Asterisk servers:
app_securax_cpuinfo.c
app_securax_serverload.c
securax_utils.c
securax_utils.h

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Copy these files in /usr/src/asterisk/asterisk/apps/ on both Origination server and Test


server.
4. Changing the applications Makefile
Now Asterisk has to be told that these modules really exits and they have to be loaded. For
this purpose you have to change the Makefile in /usr/src/asterisk/asterisk/apps/. Open the
Makefile that you have downloaded and copy the bolded lines to their corresponding places
in the original Makefile.
So you have to add these two modules in the beginning in APPS app_securax_serverload.so and app_securax_cpuinfo.so here is how this part of your
Makefile has to look after you change it.

Further on you have to add the following module definitions (this is used when you compile
Asterisk).

http://www.asteriskguru.com/tutorials/astertest.html

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securax_utils.o: app_securax_utils.c
$(CC) -pipe $(CFLAGS) -c -o app_securax_utils.o app_securax_utils.c

app_securax_cpuinfo.o: app_securax_cpuinfo.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o
app_securax_cpuinfo.oapp_securax_cpuinfo.c

app_securax_cpuinfo.so: app_securax_cpuinfo.o
$(CC) -pipe $(CFLAGS) -c -o securax_utils.o securax_utils.c
$(CC) $(SOLINK) -o $@ $< securax_utils.o

app_securax_serverload.o: app_securax_serverload.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c o app_securax_serverload.o
app_securax_serverload.c

app_securax_serverload.so: app_securax_serverload.o
$(CC) -pipe $(CFLAGS) -c -o securax_utils.o securax_utils.c
$(CC) $(SOLINK) -o $@ $< securax_utils.o

Here is how this part of the Makefile should look after you change it:

You have to change this Makefile in both Origination and Test server. In order these
changes in the Makefile to take effect you have to recompile the Asterisk on both of the
servers.
5.Re-compiling Asterisk
Go to the Asterisk source - /usr/src/asterisk/asterisk and recompile it.
cd /usr/src/asterisk/asterisk
make
make install
Now your Asterisk servers have the modules implemented. We have now to add just some
more things in the managers.conf and securax_serverload.conf files (/etc/asterisk)
6.Configuring securax_serverload.conf
This file also has to be downloaded to /etc/asterisk/. You can also create file with same
name in the mentioned directory and write in the data needed. Here is how you can do it:
cd /etc/asterisk
vi securax_serverload.conf
[general]
push_level=1
This file has to exist again on both Origination and Test servers.

7.Configuring managers.conf
In this file which is placed in /etc/asterisk you have to create a user on the Origination
server and one on the Test server, so you should have users on both servers in
managers.conf. When you start the application and before beginning a test you will be
prompted to provide username and password for each server (Origination and Test). This is
username and password for the relevant user in /etc/asterisk/managers.conf. Below you
can see my configuration.

Above the [general] account is disabled by default i.e. enabled = no. So first you have to
change this to yes. Then you have to create an account.

http://www.asteriskguru.com/tutorials/astertest.html

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[test]
this is the username
secret=test
the password is set to test
read=system,call,log,verbose,command,agent,user
write=system,call,log,verbose,command,agent,user
this is authorization level for the user for the various classes

You need to add some users and extensions to servers now.


8.Configuring iax.conf
Astertest uses some certain users for the test. Here is what you have to add to Asterisk Test
servers, no specified users are needed for the Origination server. Here are the users you
have to add to /etc/asterisk/iax.conf on the server you are going to test.

Here is the complete list of all users you can have. For every audio codec format you want to
test you must have separate user.
[testg723]
type=friend
host=dynamic
context=astertest
disallow=all
allow=g723
[testgsm]
type=friend
host=dynamic
context=astertest
disallow=all
allow=gsm
[testulaw]
type=friend
host=dynamic
context=astertest
disallow=all
allow=ulaw
[testalaw]
type=friend
host=dynamic
context=astertest
disallow=all
allow=alaw
[testg726]
type=friend
host=dynamic
context=astertest
disallow=all
allow=g726
[testadpcm]
type=friend
host=dynamic
context=astertest
disallow=all
allow=adpcm
[testslin]
type=friend
host=dynamic
context=astertest
disallow=all
allow=slin

http://www.asteriskguru.com/tutorials/astertest.html

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[testlpc10]
type=friend
host=dynamic
context=astertest
disallow=all
allow=lpc10
[testg729]
type=friend
host=dynamic
context=astertest
disallow=all
allow=g729
[testspeex]
type=friend
host=dynamic
context=astertest
disallow=all
allow=speex
[testilbc]
type=friend
host=dynamic
context=astertest
disallow=all
allow=ilbc

9. Configuring extensions.conf
Astertest also needs some certain extensions on both the Origination and the Test server.
This file is in /etc/asterisk/.
Origination Server
This context and extensions have to be added in among your extensions.

Test Server
The following context has to be added in extensions.conf of your Test server.

Now you have all the things you need to start the test. Restart Asterisk on both of the servers
and then enter the command line (CLI). To make sure that the new modules are working
check if they are loaded. Type in CLI show modules and if app_securax_serverload.so
and app_securax_cpuinfo has a corresponding 1 that means they are loaded. Another way
to check this is to type show modules like app_securax this will again show you the
modules with that name. Every module will have value 0 if it is not loaded and 1 if it is
loaded. So if everything seems correct we may start the test itself, and if you have some
problems check whether you followed precisely the steps above.
As you see above you need to have separate extensions for the different users, so as in my
file not all possible extensions are used, here is the full list of extensions for all the users.
exten => test02g723,1,Answer();
exten => test02g723,2,Playback(ast-test-g723);
exten => test02g723,3,Hangup();
exten => test02gsm,1,Answer();
exten => test02gsm,2,Playback(ast-test-gsm);
exten => test02gsm,3,Hangup();

http://www.asteriskguru.com/tutorials/astertest.html

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exten => test02ulaw,1,Answer();
exten => test02ulaw,2,Playback(ast-test-ul);
exten => test02ulaw,3,Hangup();
exten => test02alaw,1,Answer();
exten => test02alaw,2,Playback(ast-test-alaw);
exten => test02alaw,3,Hangup();
exten => test02g726,1,Answer();
exten => test02g726,2,Playback(ast-test-g726);
exten => test02g726,3,Hangup();
exten => test02adpcm,1,Answer();
exten => test02adpcm,2,Playback(ast-test-adpcm);
exten => test02adpcm,3,Hangup();
exten => test02slin,1,Answer();
exten => test02slin,2,Playback(ast-test-siln);
exten => test02siln,3,Hangup();
exten => test02lpc10,1,Answer();
exten => test02lpc10,2,Playback(ast-test-lpc10);
exten => test02lpc10,3,Hangup();
exten => test02g729,1,Answer();
exten => test02g729,2,Playback(ast-test-g729);
exten => test02g729,3,Hangup();
exten => test02speex,1,Answer();
exten => test02speex,2,Playback(ast-test-speex);
exten => test02speex,3,Hangup();
exten => test02ilbc,1,Answer();
exten => test02ilbc,2,Playback(ast-test-ilbc);
exten => test02ilbc,3,Hangup();

10. Starting Astertest


1.Run the application - astertest.exe

Here you can start new test, make again an existing test or edit an existing test. Choose
Start a new test and click Next.
2.Choose type and name for the test

Type the name you want to give to the test in the Name field at the top and choose type
(answer, playback or meetme) for the test. When done click Next.
3.Provide information for the servers

http://www.asteriskguru.com/tutorials/astertest.html

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In the Origination server part type the IP address for the Origination server, leave the
port as by default 5038. Then type the username and password that you created in
managers.conf on the corresponding Asterisk server.
In the Test server part type again the IP of the Asterisk server that you are going to test
and the username and password for the user you created in managers.conf on the
relevant server. When done click next to proceed.
4.Final adjustments for the test

On the top check the amount of calls you want to do while testing test till overload of
the tested server or test certain amount of calls. I will run the test for a hundred calls.
Speed is the speed of origination for the calls. Normal is approximately 1 call in a
second.
Protocol is the protocol type that asterisk will use. As we registered our users in
iax.conf we have to use IAX2 protocol for the test.
In Transcoding you have to choose codec for the origination server and for the tested
server. There will be no transcoding if you choose the same codecs for original and
resulting codec.
At the bottom check Send RTP this way the media file will be send. If you do not
check it you will have just answer in the test, without real voice pressure over the server.
When done click Finish to start the test.
5.Testing

http://www.asteriskguru.com/tutorials/astertest.html

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This is the test ready to be started, just click Start to run the test.

At the upper part you can see information for the origination server calls at the moment
(on the small graph on the left), CPU load (the graph on the right, showing Idle load,
system load, user load), and below more information for the calls made and channels
used at the moment.
At the bottom part you see the same information for the test server.
While the test is executed you can magnify the CPU load graph. You can do this by
clicking left mouse button and move left to right. Here is a magnified graph.

When you again click the left mouse button and move the mouse from right to left at the
same time and the graph will be minimized to its normal size.

When the test is done the calls reach the certain amount you wanted or the server is
overloaded the test is done. Then the calls are unloaded. Here you see a progress bar
while all the calls are hanging up.

Then you can see the graph from the test.

You now have the test and when you run again Astertest, you can redo the test without
doing the settings again (step 1 to 4), or you can also change a little bit the test for
example increase or decrease the amount of calls, change the codec, etc.

http://www.asteriskguru.com/tutorials/astertest.html

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User Comments
Bichu (uy4m91vgbnc at gmail dot com)
09 October 2015 17:46:20
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nusrat (nusrat_vonair at yahoo dot com)
10 February 2009 07:37:13
If i use sipp foe originating the calls is it possible?
jerry (haro1981 at 163 dot com)
19 February 2008 07:31:31
I\\\'m trying to compile it on Asterisk 1.2.18, on redhat9.0 linux.
there are missing astertest-rtp.gsm file & somewrong in Re-compiling.
make[1]: *** No rule to make target `app_qcall.so\\\', needed by `all\\\'. Stop.
make[1]: Leaving directory `/usr/src/asterisk-1.2.18/apps\\\'
make: *** [subdirs] Error 1
Kevin Zhong (jzho008 at yahoo dot co dot nz)
21 February 2007 04:18:09
I set up the original and test server by following the instruction. but while I start
to astertest on windows, I got "No Route to Host" error, what might be the
problem, and as above,the original does not make any call.
Need advice.
Many thanks,
BoneyM Thomas (boneymtom at rediffmail dot com)
09 February 2007 05:41:35
This tool works bit hard to get it running in the first place. The only stress test
tool i could find for free. It is worth the time sent setting it up.
Seb (sebb dot clarke at gmail dot com)
18 October 2006 17:53:55
Managed to get it to compile with the user comments on this page but, like
almost everyone else, the orgination server does not make any calls.
Looks like its pretty broken
Anand Kumar Gupta (kumaranandgupta at gmail dot com)
16 October 2006 12:28:43
HI ALL, THERE IS SOME MISTAKES IN THE ABOVE DOCUMENT, SO PLEASE
CORRECT IT.
1.) app_securax_cpuinfo.o: app_securax_cpuinfo.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o
app_securax_cpuinfo.oapp_securax_cpuinfo.c
THE ABOVE LINE WILL BE
app_securax_cpuinfo.o: app_securax_cpuinfo.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_securax_cpuinfo.o
app_securax_cpuinfo.c
THERE WILL BE A SPACE BETWEEN app_securax_cpuinfo.o and
app_securax_cpuinfo.c IN THE LAST LINE OF THE ABOVE LINE.
2.) IF YOU ARE GETING THE ERROR IN THE LAST FEW LINES --"You must
include stdio.h before file.h"
OPEN THE BELOW FILE WITH VI ONE BY ONE
app_securax_cpuinfo.c
app_securax_serverload.c
securax_utils.c
AND INCLUDE LINE #include <stdio.h> JUST BEFORE #include
<asterisk/file.h> AND DELETE THE #include <stdio.h> after the #include

http://www.asteriskguru.com/tutorials/astertest.html

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<asterisk/file.h> to avoid repetition of the line.
Best Regards,
Anand Kumar Gupta
Product Engg.
Troop Software Pvt. Ltd.
RuleKinG (ruleking at gmail dot com)
16 August 2006 15:47:28
I tried everything, different asterisk versions, all kind of configs but the orginating
server is not making any calls. Because it is not open source i have no idea
what is happening. Does anybody has a clue at least how to debug this...?
Roeland (roelandwerring at hotmail dot com)
14 August 2006 14:04:30
i changed the Makefile cause the files these day are not called
app_securax_utils.c but securax_utils.c :$:) Sorry for bothering..
Roeland (roelandwerring at hotmail dot com)
14 August 2006 13:14:59
I'm trying to compile it on Asterisk 1.2.10, on debian linux.
Postgresql is installed. I have the feeling it has something to do with my
compiler settings, although asterisk compiles fine without Astertest. There are
no files missing, i followed the instruction step by step.
anyone...? tia
make[1]: Entering directory `/usr/src/asterisk-1.2.10/apps'
gcc -pipe -I/usr/local/pgsql/include -pipe -Wall -Wstrict-prototypes -Wmissingprototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-framepointer -fPIC -c
gcc: no input files
make[1]: *** [app_securax_serverload.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-1.2.10/apps'
make: *** [subdirs] Error 1
eyasterisk (eyasterisk at gmail dot com)
29 July 2006 09:33:17
Hi,
I installed the tool and performed the steps given in the tutorials.
But i couldnt find any astertest-rtp.gsm file.When i usedast-test-gsm.gsm,no
calls are orginating from the server.Please help me if anybody suceedded in
the test..

Thanks
vivek (kumarvivek_24 at sify dot com)
24 July 2006 08:37:22
In our testing, origination server doesn't make any calls. What might be the
problem?
neo (job_ppa at yahoo dot co dot in)
12 July 2006 12:24:02
I want to test the trancoding for amr in asterisk. Is their any tool that test the
trancoding for amr in asterisk. This tool doesn't support for amr....
Anthony Ho (anthonyh at mcsb dot com)
30 June 2006 03:15:45
Hey guys by the way you all know where can find the astertest-rtp.gsm file? I
couldn't get it from http://astertest.com/downloads/ . Can i use other gsm like
ast-test-gsm.gsm files to subtitute the astertest-rtp.gsm? Thankz
Anthony Ho (anthonyh at mcsb dot com)
28 June 2006 12:32:44
gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissingdeclarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 march=i686 -fomit-frame-pointer -fPIC -c -o securax_utils.o securax_utils.c
gcc -shared -Xlinker -x -o app_securax_serverload.so
app_securax_serverload.o securax_utils.o -L/usr/local/pgsql/lib -lpq
/usr/bin/ld: cannot find -lpq
collect2: ld returned 1 exit status
make[1]: *** [app_securax_serverload.so] Error 1

All these can be eliminated by installing the postgresql's Devel packages. I've
done that and it compiled and loaded successfully.
Adam (ajlong at worldlink dot net)
11 June 2006 19:44:13
Edit the file called app_securax_serverload.c
add in the line
#include <stdio.h>
at the top of the group of #include statements or at least before line 17
Recompile... you may have to do this with the other source files as well
boneym (boneymtom at rediffmail dot com)
25 May 2006 08:12:53
Hi ,
i am running a asterisk 1.2.7.1 on fedora 4. Asterisk is workign fine but i get the
following error while compiling astertest. i ahev installed postgresql 8.1 too. If
anyone can point out the mistake it would have been very helpful . I am
including my console output.
[root@asteriskServer apps]# make
cc -pipe -I/usr/local/pgsql/include -fPIC -c -o app_securax_serverload.o
app_securax_serverload.c

http://www.asteriskguru.com/tutorials/astertest.html

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In file included from /usr/include/asterisk/manager.h:28,
from app_securax_serverload.c:15:
/usr/include/asterisk/lock.h: In function &#8216;ast_mutex_init&#8217;:
/usr/include/asterisk/lock.h:525: error:
&#8216;PTHREAD_MUTEX_RECURSIVE&#8217; undeclared (first use in this
function)
/usr/include/asterisk/lock.h:525: error: (Each undeclared identifier is reported
only once
/usr/include/asterisk/lock.h:525: error: for each function it appears in.)
In file included from app_securax_serverload.c:17:
/usr/include/asterisk/file.h:27:2: error: #error You must include stdio.h before
file.h!
In file included from app_securax_serverload.c:17:
/usr/include/asterisk/file.h: At top level:
/usr/include/asterisk/file.h:56: error: syntax error before &#8216;*&#8217; token
/usr/include/asterisk/file.h:57: error: syntax error before &#8216;*&#8217; token
app_securax_serverload.c: In function &#8216;scx_load&#8217;:
app_securax_serverload.c:97: warning: passing argument 3 of
&#8216;ast_pthread_create_stack&#8217; from incompatible pointer type
make: *** [app_securax_serverload.o] Error 1

Thanks
Marcus (marcus_pocus at hotmail dot com)
15 May 2006 18:41:59
Running:
Asterisk 1.2.7.1
CentOS4.3
Compilation - all ok...
Error: receiving this error message when executing asterisk...
May 14 12:34:27 VERBOSE[8469] logger.c: [app_securax_cpuinfo.so]May 14
12:34:27 WARNING[8469] loader.c:
/usr/lib/asterisk/modules/app_securax_cpuinfo.so: undefined symbol:
scx_get_cpuinfo
May 14 12:34:27 WARNING[8469] loader.c: Loading module
app_securax_cpuinfo.so failed!
any clue ??
Adam L (ajlong at worldlink dot net)
25 April 2006 06:42:47
I've gotten as far as compiling (must have postres SQL dev libs installed, thats
the -lpq error issue) and when running the tests both asterisk servers show the
management user logging in properly, and then the graph's start scrolling on
the windows test client, but no calls are originated at all... very strange... any
ideas?
Kim (autofolder at gmail dot com)
21 April 2006 15:52:03
Is there anybody succeed in test recently?
henry otasowere (henry at eportelnet dot com)
18 April 2006 03:49:51
hello,
Im new to linux and Asterisk. I needed a help. I have a Lucent Max TNT and
neew to configure it with Asterisk@Home 2.8, can any one help me out on what
to do.
Regards
Gian Luca (au_perth12 at yahoo dot it)
13 April 2006 16:11:09
Hi Jane i think i have the same problem like you.
How did you solve it?
Other does this stress-tool work?
make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.7/pbx'
/bin/sh: line 1: curl-config: command not found
make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.7/apps'
gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissingdeclarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o
securax_utils.o securax_utils.c
gcc -shared -Xlinker -x -o app_securax_serverload.so
app_securax_serverload.o securax_utils.o -L/usr/local/pgsql/lib -lpq
/usr/bin/ld: cannot find -lpq
collect2: ld returned 1 exit status
make[1]: *** [app_securax_serverload.so] Error 1
make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.7/apps'
make: *** [subdirs] Error 1
Gian Luca
Jane (yanguilin1982 at yahoo dot com dot cn)
17 March 2006 02:19:15
Why there should be two * servers? How about on * server?
Can anyone tell me why the original server doesn't make any calls?
Thank you very much.
Jane (yanguilin1982 at yahoo dot com dot cn)
10 March 2006 04:33:07
Does Astest work with Asterisk-1.2.4? When I recompile it with Asterisk-1.2.4,I
got the following info:
gcc -pipe -I/usr/local/pgsql/include -pipe -Wall -Wstrict-prototypes -Wmissingprototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -fPIC -c -o
app_securax_serverload.o app_securax_serverload.c

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app_securax_serverload.c: In function `scx_load_config':
app_securax_serverload.c:73: warning: implicit declaration of function
`ast_load'
app_securax_serverload.c:73: warning: initialization makes pointer from integer
without a cast
app_securax_serverload.c:84: warning: implicit declaration of function
`ast_destroy'
app_securax_serverload.c: In function `scx_load':
app_securax_serverload.c:96: warning: passing arg 3 of
`ast_pthread_create_stack' from incompatible pointer type
gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissingdeclarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 march=i686 -fomit-frame-pointer -fPIC -c -o securax_utils.o securax_utils.c
gcc -shared -Xlinker -x -o app_securax_serverload.so
app_securax_serverload.o securax_utils.o -L/usr/local/pgsql/lib -lpq
/usr/bin/ld: cannot find -lpq
collect2: ld returned 1 exit status
make[1]: *** [app_securax_serverload.so] Error 1
Any advice will be really appreciated!
Tang Luong (xuantang09 at yahoo dot com)
24 February 2006 07:12:13
In our testing, origination server doesn't make any calls. What might be the
problem?
Tang Luong (xuantang09 at yahoo dot com)
23 February 2006 11:31:01
In our testing, origination server doesn't make any calls. What might be the
problem?
Tang Luong (xuantang09 at yahoo dot com)
22 February 2006 07:08:01
I builded success 2 files app_securax_serverload.so and
app_securax_cpuinfo.so but when start asterisk this 2 module don't load.
http://www.asteriskguru.com/tutorials/astertest.html
$asterisk -gvvvv -c
display :
[app_securax_cpuinfo.so] => (Securax cpuinfo manager function)
== Manager registered action CPUInfo
Feb 22 01:04:55 WARNING[25502]: app_securax_cpuinfo.c:71 scx_load:
Securax cpuinfo manager function loaded.
[app_securax_serverload.so]Feb 22 01:04:55 WARNING[25502]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_securax_serverload.so:
undefined symbol: ast_load
Feb 22 01:04:55 WARNING[25502]: loader.c:554 load_modules: Loading
module app_securax_serverload.so failed!
Ouch ... error while writing audio data: : Broken pipe
Warning, flexibel rate not heavily tested!
please help me fix it!!!
thanks
faraz majid (faraz dot m dot ali at gmail dot com)
23 January 2006 12:15:50
Well i managed to load the modules on Asterisk version 1.2.1. edit the file
apps/app_securax_serverload.c , on line 75 change : 'ast_load' to
'ast_config_load' & similarly on line 86 change 'ast_destroy' to
ast_config_destroy' save the changes and recomplie asterisk... u r done..
For testing SIP calls:
---------------------In /etc/asterisk/sip.conf
under the [general] section set as follows
[general]
context=astertest
allowguest=yes
all is done,
if u run into trouble then u can contact me
allen (qsun at mdc dot cn)
09 January 2006 03:29:23
useless!!!!!
allen (qsun at mdc dot cn)
31 December 2005 06:53:58
mechanix <dmitry dot muntean at gmail dot com>
11 November 2005 16:20:53
does it work with asterisk 1.2beta?
i get following during asterisk startup:
[app_securax_serverload.so]Nov 11 17:26:18 WARNING[16466]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_securax_serverload.so:
undefined symbol: ast_load
Nov 11 17:26:18 WARNING[16466]: loader.c:554 load_modules: Loading
module app_securax_serverload.so failed!
Ouch ... error while writing audio data: : Broken pipe

yes,we meet the same problem as above,can someone te us the reason.


we all followed by the steps as shown in this article.
thanks a lot.
Geongo (geongo at yahoo dot com)

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28 December 2005 20:36:18
What is the maximum number of calls that you achieved and the maximum
Calls per Second Rate? What was your hardware configuration for the servers?
mechanix (dmitry dot muntean at gmail dot com)
11 November 2005 16:20:53
does it work with asterisk 1.2beta?
i get following during asterisk startup:
[app_securax_serverload.so]Nov 11 17:26:18 WARNING[16466]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_securax_serverload.so:
undefined symbol: ast_load
Nov 11 17:26:18 WARNING[16466]: loader.c:554 load_modules: Loading
module app_securax_serverload.so failed!
Ouch ... error while writing audio data: : Broken pipe
spyke (cosimo dot fadda at gmail dot com)
27 October 2005 16:09:06
It is the same to me; I'm using sip instead of iax and I'm not able to choose the
codec translaction.
C.
opeVoce (support at vocee dot com)
19 October 2005 00:43:39
Yes, we exactly followed the instructions. In manager.conf, we change
'push_level=1' to 'push_rate=1'. We saw manager registered on both servers.
We saw login, status commands in both asterisks. We saw 'cpuinfo
commands' only on the test server. There is nothing else.
I will send you screenshots from email.
ivan (support at asteriskguru dot com)
18 October 2005 11:07:31
Do you follow exactly the steps above. Did you tested it on some other pc
successfully. What does the test shows. Can you send some screenshots and
description of what you are doing.
opeVoce (support at vocee dot com)
18 October 2005 07:01:43
In our testing, origination server doesn't make any calls. What might be the
problem?
ivan (support at asteriskguru dot com)
20 September 2005 10:41:50
All the screenshots above in the tutorial are results from a real test
Senad (Senad at bicom dot us)
16 September 2005 19:32:23
Hi,
Have you any test results published yet?

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