search for: 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() however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: ap...

...f_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 *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.xml PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_1.pcap UNKNOWN *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/dtmf_2833_1_noend.pcap UNKNOWN *>* /asterisk/tr...

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:...

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...

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 done with both signaling and media like this: SIPp <---> SUT (asterisk 13) <---> Asterisk box echoing media I checked bandwidth first and got 930 Mb/s on each leg (from SIPp to SU...

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 in...

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 easier, we created Sippy Cup. I wanted to announce it here in the hopes that it makes your lives easier as well. Sippy C...

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???????????? |...

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...

...omanian language - we plan to provide an English version soon). This article is describing a method to be used for obtaining the maximum number of SIP simultaneous calls an Asterisk server could process safely (meaning no errors/maintain control of the machine and without RTP frame drops) We used SIPP (with modified uas and uac_pcap scenarios) + 2 scripts for controlling the test (one is running on the tested Asterisk server - start-test.sh, for data collection and load analysis and the other is running on the SIPP+Asterisk testing machine, for call quality control and SIPP instance control - si...

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 ---&gt...

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 to...

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: [Asterisk-Users] SIPP...

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 '&...

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 900...

Hi all, I'm trying to use SIPP (http://sipp.sourceforge.net/) to stress-test our asterisk installation. We have a very simple dialplan that uses FastAgi. I'm finding that all calls to "GET VARIABLE" from the FastAgi are returning null when the dialplan is invoked from sipp -- and they work fine when invoked fro...

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] WARNIN...

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 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 server(192.168.1.100)------------------------------------------->UAS:192.168.1.101 Registering: UAC:192...

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