similar to: Article - a method on how to evaluate an Asterisk server

Hi friends... I am trying to test my voicemail with Asterisk using SIPP (SIPP is running in Debian Squezze and Asterisk is running in OpenSuSE-11.1), the command that I use is: sipp -sn uac_pcap -l 1 -m 1 -s 55 -trace_err 192.168.13.6 But, If I use the file g711a.pcap included in the sources of sipp or if use some file captured for me the result is the same ---> error ... the message in

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()

Hi Guys! I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could handle in production so following is my senario. [sipp_client]---------------[Asterisk]----------------[sipp_server] sipp_client ./sipp -sf uac_pcap.xml -d 100000 -i 172.30.254.211 -s 2000 172.30.1.47 -l 1000 -r 250 -rp 5000 -m 1000 sipp_server ./sipp -sn uas -i 172.30.245.208 In above if i set -r

Hi everybody, got it from svn: dtmf_2833_1.pcap */asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN *>*

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

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 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, 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 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

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 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

Hi, We have a scenario wherein the endpoint needs to send a 600 Busy Everywhere after receiving an INVITE. I am using SIPp as this end point. SIPp is configured as UE2. Now when UE1 calls UE2 (SIPp) receives the INVITE and responds with a 600 Busy Everywhere. But when Asterisk receives this 600 response it sends out a 486 Busy Here to UE1. Ideally Asterisk should be relaying the 600

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

hi, i'm evaluating performance of centos7 i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 with 500calls (7sec alaw, simple dialplan, pass trough - sipp generators/asterisk receiver with answer/playback) scenario - sipp generators - asterisk - asterisk receiver (i wrote ansible scenario for this if you are interested) then i reinstalled system to centos7 x86_64/distro

Hello, I am trying to run load on asterisk server(version 1.8.7.1) through SIPp tool for the voicemail() application. But I am facing a lot of problems. I tried running 1000 calls?from SIPp for 100 subscribers (10 messages for each subscriber). I am using odbc storage for the messages. Following warnings/errors are coming on the asterisk server: Jan 11 11:30:49] WARNING[22924] app.c:

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 all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i

Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line

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

Has anyone created any docker images I might be able to use on EC2 for load testing an asterisk platform? I started an instance this morning and was about to load sipp and other tools, and then thought surely someone must have done this already. I'd like to hammer a platform we have created with multiple EC2 images until it breaks, to test capacity. Cheers, j