similar to: Asterisk and Sipp

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

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:

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

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()

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

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 ?

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, 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

Hi ALL, I have install asterisk 1.8.2.3 on my ubuntu 10.x machine with 512 MB memory. Now i am running sipp tester to check performance but at some point in running test my asterisk got freez its doing nothing but i can run commands on CLI, But it doesn't accepting new request this time. following test result of sipp. I am playing default music and hold for sipp test. sipp -sn uac -d

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 from a softphone on the same machine, for example. Does anyone have

Hi all, My experiment scenario is like this: SIPp Uac -----------------------------> ASTERISK SERVER---------------------------------->SIPp uas 1. when i had registered bob with this command ./sipp -sf register_client.xml -inf register1.csv -i 192.168.1.6:5060 192.168.1.6 -p 5061 -m 10000 it has registered.... If i want to register another client alice with same command

Hello, I have asterisk 1.4.4 running with anonymous sip calls enabled and I am testing the box for load using sipp with something like this - sipp -sn uac -s 10 -d 60000 -i 192.168.1.49 -l 110 -r 5 -trace_err 192.168.1.50 Asterisk picks up the call and runs a test php-agi file that plays a .gsm file. As soon as the number of active calls reaches 99, asterisk starts Declining further calls. I

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

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

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

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 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

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:

I have the following scenario: g729 gsm UAS <-----------> * <-----------> UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see 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