Comments on: Using sipp to stress test your asterisk 1.4 pbx system

By: Gopal

Gopal — Thu, 27 Jan 2011 16:32:10 +0000

I tried to execute as per your tutorial. I created one extension 2001 in Asterisk and one VoIP trunk sipp and I executed as instructed,
./sipp -sn uac -d 20000 -s 2001 192.168.0.17 -l 30 -trace_err

but while end of the result, it results to this error,

Discarding message which can’t be mapped to a known SIPp call:

Please advice what was wrong. Thanks in advance.

By: Bocah Ingusan

Bocah Ingusan — Tue, 11 May 2010 08:11:12 +0000

is SIPp must installed in VoIP server/ asterisk??

how do i setting extension.conf or sip.conf using freepbx?

please

By: Dago

Dago — Mon, 23 Nov 2009 14:15:48 +0000

Hi…. I’d like to add some knowledge to this article.

According to Prabhakar problem, where he get a Retransmission error…. after a lot of reading, I came to realize that enabling “nat=yes” in sip.conf file (asterisk box) would solve the problem….. and fortunately, I was right.
The explanation, not quite complete, but reasonable, is that I notice (enabling debuging for sip in Asterisk CLI and tracing Sipp errors logs), that no “100″ message (part of SIP protocol) was return form Asterisk, but that doesn’t mean that it doesn’t send it, as a matter of fact it does, but non of them reach Sipp, so…. as no 100 message is get, Sipp start restransmiting, and in some point it’ll reach the limit, so “chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 30-3832@127.0.0.1 for seqno 1 (Critical Response)” message will be outputed.
By enabling NAT this problem is solved. A deep explanation can be found in the sample sip.conf file form Asterisk (if you ran “make samples” when asterisk was compiled)

By: Dago

Dago — Wed, 18 Nov 2009 14:27:51 +0000

Hi… let me know somthing. Did you run sipp command from the same machine that runs Asterisk?

By: Asterisk SNMP with Cacti Howto upgraded for Asterisk 1.6 and Ubuntu!

Asterisk SNMP with Cacti Howto upgraded for Asterisk 1.6 and Ubuntu! — Tue, 28 Oct 2008 19:03:43 +0000

[...] let the poller do it’s work. Make some calls, or stress test your Asterisk system using our SIPP howto and watch the graphs come to life. Here’s some samples right after we created our [...]

By: Little Helper

Little Helper — Mon, 27 Oct 2008 14:40:46 +0000

Mostly the upper limit of calls is set by the limit of open files.
Check it out by typing “ulimit -a”
Or just set it to 65000 by typing “ulimit -n 65000″

greetz

By: Voip Phreak

Voip Phreak — Fri, 29 Aug 2008 18:58:55 +0000

Hi Aurobindo,

Thanks for the comments. I’ve updated the link for you at the top of the article – good catch. Hope this helps.

Thanks,
VP

By: Aurobindo

Aurobindo — Wed, 27 Aug 2008 12:09:24 +0000

Hi Voip Phreak,

this is a nice article i must say. Nice piece of info. My problem is that can you please let me know how the graphs you have generated as the link you have given here is not working. please mail me in my above address so that i can find your site easily. Hope to get the answer soon. liked your name a lot!!

Thanks
Aurobindo

By: -ab-

-ab- — Fri, 04 Apr 2008 14:31:26 +0000

You have to use sipp w/ an scenario where the SDP offers GSM.

By: shahrul

shahrul — Thu, 15 Nov 2007 13:05:14 +0000

thanks Matt for your help. but the links does not help me alot.

the 1,2,4 in the extension already corrected.. but the asterisk respond still the same.. no compatible codec.. but when i try use any GSM supported phone, it work well. it just asterisk refuse to acknowledge sipp through GSM codec..

maybe other visitors can solve this issue. tq