similar to: Chan_sip max channels limit?

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 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:

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

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

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, 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()

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 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, 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

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

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 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

http://sipp.sourceforge.net/ Anyone care to throw this at Asterisk to see what happens? I would, but I am having significant temporal shortfalls recently due to the apparent warping of the space/time continuum when I answer the phone with clients/associates. It seems that entire days pass by before I hang up... very odd, and very counter-productive to getting good Asterisk work done. JT

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 *>*

On Fri, May 4, 2012 at 6:50 PM, vaibhav srivastava <vaibhavcse30 at gmail.com>wrote: > > Hi all, > Since I want to run Samba without modifying my existing kernel. Please > tell me what are the requirements for the same. > What are the package list required in kernel before installing samba. > thanks in advance. > -- > Thanks and Regards, > Vaibhav Srivastava >

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 all, I would like to share with you an article [1] we have issued last week (sorry, currently only in Romanian 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)

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