Astertest - asterisk stress testing tool.pdf

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3/11/2015 Astertest - asterisk stress testing tool http://www.asteriskguru.com/tutorials/astertest.html 1/24 Idefisk Tools Tutorials Reviews VoIP Providers Archives Search Site 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. 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 Copy these files in /usr/src/asterisk/asterisk/apps/ on both Origination server and Test server. 4. Changing the application’s 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). Latest Headlines: T.38 faxing w ith Zoiper 2.15 is now easier than ever section: voip software Asterisk 1.4.21 Released section: Asterisk Asterisk 1.4.20 Released section: Asterisk Asterisk 1.4.20-rc2 Released section: Asterisk Asterisk 1.4.20-rc1 Now Available section: Asterisk New s Archives (older new s) Latest Tutorials: Sending Fax from Zoiper to Zoiper using T.38 added 08/Dec/2008 18:16 VMAuthenticate (dialplan application) added 01/Mar/2008 15:57 Siptronic ST-530 added 06/Nov/2007 17:57 Siemens C455 IP hardphone added 05/Nov/2007 10:24 Zoiper added 22/Oct/2007 17:53 Latest Comments: 5.0 出 5 明星 Cheapest appro... tutorial: SIP with NAT or Firewalls Jody,I may be younger at 49 than I w as a... tutorial: Asterisk Installation Hello Loving Care,Signed hard copy is ma... tutorial: RxFax (dialplan application) Hi Bill I am using TPG internet and th... tutorial: How to install and configure Wildcard TDM400p Helped a friend hsarevt some Frontenac G... tutorial: sip_poke_noanswer: Peer 'XXX' is now UNREACHABLE! Back to Tutorials

Transcript of Astertest - asterisk stress testing tool.pdf

Page 1: Astertest - asterisk stress testing tool.pdf

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Idefisk Tools Tutorials Reviews VoIP Providers Archives

Search Site

9. Astertest - asterisk stress testing tool

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

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 fromhttp://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 forexample - ast-test-al.al. You can use your own files with different name, but make surewhen you configure the extensions.conf on the Test server to use the correct audio filename 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 loadwhile testing.

3.New Asterisk Modules

The modules below must be installed on both Asterisk servers:app_securax_cpuinfo.capp_securax_serverload.csecurax_utils.csecurax_utils.h

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

4. Changing the application’s Makefile

Now Asterisk has to be told that these modules really exits and they have to be loaded. Forthis purpose you have to change the Makefile in /usr/src/asterisk/asterisk/apps/. Open theMakefile that you have downloaded and copy the bolded lines to their corresponding placesin 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 yourMakefile has to look after you change it.

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

Latest Headlines:

T.38 faxing w ith Zoiper 2.15 is now

easier than ever

section: voip software

Asterisk 1.4.21 Released

section: Asterisk

Asterisk 1.4.20 Released

section: Asterisk

Asterisk 1.4.20-rc2 Released

section: Asterisk

Asterisk 1.4.20-rc1 Now Available

section: Asterisk

New s Archives (older new s)

Latest Tutorials:

Sending Fax from Zoiper to Zoiper

using T.38

added 08/Dec/2008 18:16

VMAuthenticate (dialplan application)

added 01/Mar/2008 15:57

Siptronic ST-530

added 06/Nov/2007 17:57

Siemens C455 IP hardphone

added 05/Nov/2007 10:24

Zoiper

added 22/Oct/2007 17:53

Latest Comments:

5.0 出 5 明星 Cheapest appro...

tutorial: SIP with NAT or Firewalls

Jody,I may be younger at 49 than I

w as a...

tutorial: Asterisk Installation

Hello Loving Care,Signed hard copy is

ma...

tutorial: RxFax (dialplan application)

Hi Bill I am using TPG internet and th...

tutorial: How to install and configure

Wildcard TDM400p

Helped a friend hsarevt some

Frontenac G...

tutorial: sip_poke_noanswer: Peer 'XXX'

is now UNREACHABLE!

Back to Tutorials

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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 -oapp_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.oapp_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 thesechanges 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/asteriskmake

make install

Now your Asterisk servers have the modules implemented. We have now to add just somemore 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 samename in the mentioned directory and write in the data needed. Here is how you can do it:

cd /etc/asteriskvi 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 Originationserver 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 beprompted 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 youcan 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.

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[test]

this is the username

secret=test

the password is set to test

read=system,call,log,verbose,command,agent,userwrite=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 Testservers, 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 totest you must have separate user.

[testg723]

type=friendhost=dynamic

context=astertestdisallow=all

allow=g723

[testgsm]

type=friendhost=dynamic

context=astertestdisallow=allallow=gsm

[testulaw]type=friendhost=dynamic

context=astertestdisallow=allallow=ulaw

[testalaw]type=friendhost=dynamic

context=astertestdisallow=allallow=alaw

[testg726]type=friendhost=dynamic

context=astertestdisallow=allallow=g726

[testadpcm]type=friendhost=dynamiccontext=astertest

disallow=allallow=adpcm

[testslin]type=friendhost=dynamiccontext=astertest

disallow=allallow=slin

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[testlpc10]type=friendhost=dynamic

context=astertestdisallow=allallow=lpc10

[testg729]type=friendhost=dynamic

context=astertestdisallow=allallow=g729

[testspeex]type=friendhost=dynamiccontext=astertest

disallow=allallow=speex

[testilbc]type=friendhost=dynamiccontext=astertest

disallow=allallow=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 serversand 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 wayto check this is to type show modules like app_securax – this will again show you themodules 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 someproblems 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();

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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

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In the “Origination server” part type the IP address for the Origination server, leave theport as by default – 5038. Then type the username and password that you created inmanagers.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 therelevant 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 ofthe 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 asecond.

Protocol – is the protocol type that asterisk will use. As we registered our users iniax.conf we have to use IAX2 protocol for the test.

In Transcoding you have to choose codec for the origination server and for the testedserver. There will be no transcoding if you choose the same codecs for original andresulting 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

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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 channelsused 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 byclicking 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 thesame 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 isoverloaded the test is done. Then the calls are unloaded. Here you see a progress barwhile 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 withoutdoing the settings again (step 1 to 4), or you can also change a little bit the test – forexample increase or decrease the amount of calls, change the codec, etc.

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User Comments

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nusrat (nusrat_vonair at yahoo dot com)10 February 2009 07:37:13If i use sipp foe originating the calls is it possible?

jerry (haro1981 at 163 dot com)19 February 2008 07:31:31I\\\'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 04:18:09I set up the original and test server by following the instruction. but while I startto astertest on windows, I got "No Route to Host" error, what might be theproblem, and as above,the original does not make any call.Need advice.

Many thanks,

BoneyM Thomas (boneymtom at rediffmail dot com)09 February 2007 05:41:35This tool works bit hard to get it running in the first place. The only stress testtool 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 17:53:55Managed 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 12:28:43HI ALL, THERE IS SOME MISTAKES IN THE ABOVE DOCUMENT, SO PLEASECORRECT IT.

1.) app_securax_cpuinfo.o: app_securax_cpuinfo.c$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -oapp_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 andapp_securax_cpuinfo.c IN THE LAST LINE OF THE ABOVE LINE.

2.) IF YOU ARE GETING THE ERROR IN THE LAST FEW LINES --"You mustinclude stdio.h before file.h"

OPEN THE BELOW FILE WITH VI ONE BY ONE

app_securax_cpuinfo.capp_securax_serverload.csecurax_utils.c

AND INCLUDE LINE #include <stdio.h> JUST BEFORE #include<asterisk/file.h> AND DELETE THE #include <stdio.h> after the #include

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<asterisk/file.h> to avoid repetition of the line.

Best Regards,

Anand Kumar GuptaProduct Engg.Troop Software Pvt. Ltd.

RuleKinG (ruleking at gmail dot com)16 August 2006 15:47:28I tried everything, different asterisk versions, all kind of configs but the orginatingserver is not making any calls. Because it is not open source i have no ideawhat is happening. Does anybody has a clue at least how to debug this...?

Roeland (roelandwerring at hotmail dot com)14 August 2006 14:04:30i changed the Makefile cause the files these day are not calledapp_securax_utils.c but securax_utils.c :$:) Sorry for bothering..

Roeland (roelandwerring at hotmail dot com)14 August 2006 13:14:59I'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 mycompiler settings, although asterisk compiles fine without Astertest. There areno 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 -Wmissing-

prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c gcc: no input filesmake[1]: *** [app_securax_serverload.o] Error 1make[1]: Leaving directory `/usr/src/asterisk-1.2.10/apps'make: *** [subdirs] Error 1

eyasterisk (eyasterisk at gmail dot com)

29 July 2006 09:33:17Hi,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,nocalls are orginating from the server.Please help me if anybody suceedded inthe test..

Thanks

vivek (kumarvivek_24 at sify dot com)24 July 2006 08:37:22In our testing, origination server doesn't make any calls. What might be theproblem?

neo (job_ppa at yahoo dot co dot in)12 July 2006 12:24:02I 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 03:15:45Hey guys by the way you all know where can find the astertest-rtp.gsm file? Icouldn't get it from http://astertest.com/downloads/ . Can i use other gsm likeast-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 -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -fPIC -c -o securax_utils.o securax_utils.cgcc -shared -Xlinker -x -o app_securax_serverload.soapp_securax_serverload.o securax_utils.o -L/usr/local/pgsql/lib -lpq/usr/bin/ld: cannot find -lpqcollect2: ld returned 1 exit statusmake[1]: *** [app_securax_serverload.so] Error 1

All these can be eliminated by installing the postgresql's Devel packages. I'vedone that and it compiled and loaded successfully.

Adam (ajlong at worldlink dot net)11 June 2006 19:44:13Edit the file called app_securax_serverload.cadd 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 thefollowing error while compiling astertest. i ahev installed postgresql 8.1 too. Ifanyone can point out the mistake it would have been very helpful . I amincluding my console output.[root@asteriskServer apps]# makecc -pipe -I/usr/local/pgsql/include -fPIC -c -o app_securax_serverload.o

app_securax_serverload.c

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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 thisfunction)/usr/include/asterisk/lock.h:525: error: (Each undeclared identifier is reportedonly 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 beforefile.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 typemake: *** [app_securax_serverload.o] Error 1

Thanks

Marcus (marcus_pocus at hotmail dot com)

15 May 2006 18:41:59Running:Asterisk 1.2.7.1CentOS4.3

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 1412:34:27 WARNING[8469] loader.c:/usr/lib/asterisk/modules/app_securax_cpuinfo.so: undefined symbol:scx_get_cpuinfoMay 14 12:34:27 WARNING[8469] loader.c: Loading moduleapp_securax_cpuinfo.so failed!

any clue ??

Adam L (ajlong at worldlink dot net)25 April 2006 06:42:47I've gotten as far as compiling (must have postres SQL dev libs installed, thatsthe -lpq error issue) and when running the tests both asterisk servers show themanagement user logging in properly, and then the graph's start scrolling onthe windows test client, but no calls are originated at all... very strange... anyideas?

Kim (autofolder at gmail dot com)21 April 2006 15:52:03Is there anybody succeed in test recently?

henry otasowere (henry at eportelnet dot com)18 April 2006 03:49:51hello,I´m new to linux and Asterisk. I needed a help. I have a Lucent Max TNT andneew 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 16:11:09Hi 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 foundmake[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.7/apps'gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -osecurax_utils.o securax_utils.c

gcc -shared -Xlinker -x -o app_securax_serverload.soapp_securax_serverload.o securax_utils.o -L/usr/local/pgsql/lib -lpq/usr/bin/ld: cannot find -lpqcollect2: ld returned 1 exit statusmake[1]: *** [app_securax_serverload.so] Error 1make[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 02:19:15Why 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 04:33:07Does Astest work with Asterisk-1.2.4? When I recompile it with Asterisk-1.2.4,Igot the following info:gcc -pipe -I/usr/local/pgsql/include -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -fPIC -c -oapp_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 integerwithout a castapp_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 typegcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -fPIC -c -o securax_utils.o securax_utils.cgcc -shared -Xlinker -x -o app_securax_serverload.soapp_securax_serverload.o securax_utils.o -L/usr/local/pgsql/lib -lpq

/usr/bin/ld: cannot find -lpqcollect2: ld returned 1 exit statusmake[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 theproblem?

Tang Luong (xuantang09 at yahoo dot com)23 February 2006 11:31:01In our testing, origination server doesn't make any calls. What might be theproblem?

Tang Luong (xuantang09 at yahoo dot com)22 February 2006 07:08:01I 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 12:15:50Well 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' toast_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=astertestallowguest=yes

all is done,

if u run into trouble then u can contact me

allen (qsun at mdc dot cn)

09 January 2006 03:29:23useless!!!!!

allen (qsun at mdc dot cn)

31 December 2005 06:53:58mechanix <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_loadNov 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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contact us at: [email protected] - asterisKGuru.com © all rights reserved | *asterisk is registered trademark of © Digium™

28 December 2005 20:36:18

What is the maximum number of calls that you achieved and the maximumCalls 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_loadNov 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 thecodec translaction.

C.

opeVoce (support at vocee dot com)

19 October 2005 00:43:39Yes, 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 11:07:31Do 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 anddescription of what you are doing.

opeVoce (support at vocee dot com)

18 October 2005 07:01:43In our testing, origination server doesn't make any calls. What might be the

problem?

ivan (support at asteriskguru dot com)

20 September 2005 10:41:50All the screenshots above in the tutorial are results from a real test

Senad (Senad at bicom dot us)

16 September 2005 19:32:23Hi,

Have you any test results published yet?

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