similar to: Testing asterisk with sipp

We are in the process of implementing realtime voicemail. I was wanting to "stress-test" the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup()

Hello, I'm stressing an Asterisk 11 platform with incoming calls from sipp 3.1. I've dedicated a context to sipp in my dialplan. Everything works OK expect that calls from sipp comes in with a CallerID set to sipp and this sipp value is stored in CDR. 1. I can change the value of the CallerID but how can I have the calls from sipp traced in CDR with a customized src field value ?

Hi, i am trying to install the Asterisk test suite on my ubuntu system , i have followed all the installlation steps as mentioned in the link ( http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/) , but when i am trying to run the script some of the test cases are PASSED and most of them are FAILED and SKIPPPED. So please help me out to do the testing correctly. The

Hello everyone, Recently we've been focusing quite heavily on making Adhearsion[0] faster. To do that, we needed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR. So to make our own lives

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

You maybe have to create a SIP user called like it is declared in your UAC/UAS xml file. I think it should be 'sipp' or something like that... -----asterisk-users-admin@lists.digium.com a ?crit : ----- Pour: <asterisk-users@lists.digium.com> De: "C. Johnson" <javadude@cedrick.net> Envoy? par: asterisk-users-admin@lists.digium.com Date: 31-05-2004 08:03 Objet: RE:

Hello, I'm running load testing sessions. My System Under Test is an asterisk 13 with 16GB, configured with maxfiles set to 400 000. This system is supposed do produce simple SIP trunking services without transcoding. The box sending call to my System Under Test is anabled with SIPp. I'm banging on a 700 concurrent calls/50 CAPS limit I would like to improve, if possible. Tests are

Hello, SIPp's PCAP play feature can replay pre-recorded audio stream towards destination (see [1]). Doc mentions tcpdump and Wireshark as tools to record such RTP streams without further details. Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/ directory. Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to 10.1.6.18:2006 1. How can you "forge" IPs

Hi there! Does anyone knows how to test Asterisk load with sipp? I am using uac.xml to call a 'playback extensions' via a SIP channel. When I increase the Call rate (about 20cps), I begin to have INVITE/200/BYE retransmissions meanwhile the RedHat box is not loaded at all (made a TOP). Where is the pb? [root@10.54.196.38 sipp]# sipp 10.54.196.32 -s 9001 -sf uac.xml -d 100 -i 10.54.196.38

Hi every one, Hello i am doing project on evaluating the sip proxy performances like asterisk, openims and opensips using the traffic generator SIPp. I am using 2 computers of same configuration as SIPp clients one as uac and other as uas... and one laptop for asterisk server...... UAC:192.168.1.99------------------------>Asterisk

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

Hi, I just installed Sipp 3.3?on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. SIpp output: ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- ? Start Time???????????? | 2013-05-20?22:53:08:637?1369083188.637273??????????? ? Last Reset

Hey I'm looking for an advanced scenario for sipp, that can be used for testing asterisk. Mainly I'm interested in making random calls between sipp pseudo-users. Did anyone try to do something like this? Or has anyone got an example scenario with working loops? Thanks

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

Hi, sipp (http://sipp.sourceforge.net/) seems to be a good app. Take a look at http://www.voip-info.org/wiki-SIPP on the wiki to have more info about it... Basically, there is scenario which are describe there and I personnally generated about 3,000,000 calls before having to restart asterisk and i placed about 90 concurrent calls. Good luck! -----asterisk-users-admin@lists.digium.com a ?crit :

Good morningI was wondering whether you could help me. I installed sipp on my Asterisk server but I don't really understand how does it fonction! Has someone ever tried it?If you can explain to me the principle, I would be extremely grateful.Thank you very much in advance. _________________________________________________________________ Lancez des recherches en toute s?curit? depuis

Hi, How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be "programmed" to send out digits after specified time to simulate users pressing menu items. If it can originate large number of calls simultaneously then it's great! Does any one have any recommendations ? Any other method to stress test an IVR call flow? with regards, raj

First I freely admit that while I can figure out most of what is happening in the .conf files I still don't fully understand how to set up something new. I am trying to use SIPp to do some testing of stuff with asterisk but I am not sure how to set up asterisk and especailly the .conf files to do this. I saw some information on the wiki but did not see how to set up the sip.conf and

Hi Barry Flanagan, Barry Flanagan <barryf-lists at flanagan.ie> schrieb am Mit, 19. Aug 11:06: > SIPP is probably what you seek. http://sipp.sourceforge.net/ > > Hope this helps. That looks pretty like what I'm looking for! Many thanks! Sincerely, Dominique Haeber

Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live production. I cannot run valgrind and I do not have the right flags set in menuselect. I can however at the dead of the night run stress tests. I want to simulate x-amount of concurrent calls to both a dtmf dialplan, which is working, as well as MoH dialplan to see if this could be the cause of crashing. How