search for: astertest

Hi all!! I've made some changes to the applications that Astertest was using to monitor the performance of the server. Now is also possible to track the bandwidth usage of the server, this has nothing to do with the executable (astertest.exe) itself but with the events that the Asterisk Manager generates. The method described in: http://www.asteriskguru.com/tut...

Hi, Did someone ever managed to make Astertest (http://www.asteriskguru.com/tutorials/astertest.html) work ? I followed all the instructions of this tutorial and corrected the mistakes pointed by the users but it still doesn't work. I can compile it and load app_securax_cpuinfo.so. When trying to load app_securax_serverload.so I have t...

...will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable relase. Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this out and complete it in time. Please paypal your contribution to sponsor@astertest.com today. Every little dollar is worth quite a lot! I fully trust that Joachim (Zoa) and his team will complete this in a good way and look forward to improved sound quality in the SIP channel. Read more here: http://www.astertest.com/forum/viewtopic.php?t=13 Thank you for your contribution!...

...t (also periodic?), but I don't know how to check this. If this top reading is an artefact, is there a way to check the actual (realtime) load? Regarding the actual processor usage for speex encoding: this report suggests my processor is indeed quite busy encoding a few speex channels: http://astertest.com/astricon_performance.ppt. Given the results in this report, I doubt the PSTN gateway will support more than 10 speex encodings. At the same time, the same processor encodes 756x756 PAL television to mpeg-4 on my mythtv box at home. Twice, leaving room for scheduled jobs. Has anyone some referen...

It there avalible quality and delay test for sip connections for asterisk. Something like to clients making a call with different codecs and measuring delay , jitter ? I know there is a Astertest but in that you need 2 asterisk mashines (which is usually hard to have). I was looking for perl/bash scripts running sip clients in a finite loop + etheral to measure packet properties , gathering logs. Is there anything like that? ps. I have hudge delays (about 1 sec ! )when calling sip-sip an...

Hi all, I'm planning to test my two Asterisk@home one is 1.5 and another is 2.5 Does any one got already "Astertest - asterisk stress testing tool" working one? I've red Asterisk Guru, http://www.asteriskguru.com/tutorials/astertest.html and after all the tutorial still remaining questions from users with problems ( in fact i didn't find any sucessfull feedback). I'm a bit afraid of doing al...

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The

...ee it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ?Am I wrong? The UAC and UAS are registering with * properly: --- sip.conf ------------------------------------------------------------ [testgsm] type=friend host=dynamic username=testgsm context=astertest canreinvite=no disallow=all allow=gsm [testg729] type=friend host=dynamic username=testg729 context=astertest canreinvite=no disallow=all allow=g729 ------------------------------------------------------------------------- -------------------------------------------------------------------------...

Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729

hello there, any one who used ASTCC in a real enviroment, or has successfully handled above 1k simultanous calls. need some evalution of ASTCC. if any one has such an experience please share it with the rest thank you Atif

>> Has anyone benchmarked Asterisk on a dedicated single versus dual >> processor machine? > > http://www.astertest.com/ > > Cheers, Philipp The test results that Philipp pointed out show some protocol comparisons that include "iax2 trunking / alaw" and "iax2 / alaw" and concludes that "IAX2 trunking is more than twice as fast as non trunking IAX." Forgive the newbie ques...

.... IAX calls can go 400+, so I test with combination 200+ SIP calls and the rest IAX and a combination of more and less SIP and IAX calls. Memory usage never goes over 256Meg, not sure why. Interesting, findings are very consistent with other performance testing that has been done over the years, Astertest and the like. HT turned on, SMB loaded in the kernel gave ~20% performance increase, BUT, using 425 + channels gave very inconsistent results, choppy audio, calls dropped, no audio, call setup time slowed. Good results below that mark, but not enough to warrant using full time. I'd rather bu...

...test with combination 200+ SIP calls and the rest IAX and a > combination of more and less SIP and IAX calls. > > Memory usage never goes over 256Meg, not sure why. > > Interesting, findings are very consistent with other performance > testing that has been done over the years, Astertest and the like. > > HT turned on, SMB loaded in the kernel gave ~20% performance increase, > BUT, using 425 + channels gave very inconsistent results, choppy > audio, calls dropped, no audio, call setup time slowed. Good results > below that mark, but not enough to warrant using full...