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Load testing via SIPp

 
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prabhakar_asterisk



Joined: 21 Sep 2007
Posts: 1

PostPosted: Fri Sep 21, 2007 6:26 am    Post subject: Load testing via SIPp Reply with quote

Hi Everyone,

I am trying to do stress testing via SIPp

this is my sip.conf
[asterisk02]
type=friend
context=testing
host=10.168.1.107(my test server)
user=sipp
canreinvite=no
allow=all

this is my extensions.ael
context testing {
1 => {
Answer();
//Background(6604);
//Wait(2);
//&queue_helpdesk(0);
Hangup();
};
};

I am getting following error in asterisk console

[09-20 12:08:53] WARNING[122]: chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 16-3351@127.0.0.1 for seqno 1 (Critical Response)
[09-20 12:08:53] WARNING[122]: chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 17-3351@127.0.0.1 for seqno 1 (Critical Response)
[09-20 12:08:53] WARNING[122]: chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 18-3351@127.0.0.1 for seqno 1 (Critical Response)
[09-20 12:08:53] WARNING[122]: chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 19-3351@127.0.0.1 for seqno 1 (Critical Response)
[09-20 12:08:54] WARNING[122]: chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 20-3351@127.0.0.1 for seqno 1 (Critical Response)
[09-20 12:08:54] WARNING[122]: chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 21-3351@127.0.0.1 for seqno 1 (Critical Response)
[09-20 12:08:54] WARNING[122]: chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 22-3351@127.0.0.1 for seqno 1 (Critical Response)
[09-20 12:08:54] WARNING[122]: chan_sip.c:2171 retrans_pkt: Maximum retries exceeded on transmission 23-3351@127.0.0.1 for seqno 1 (Critical Response)

and I am getting following message in test server using SIPp

Aborting call with Call-Id '9-3351@127.0.0.1'.
sipp: There were more errors, enable -trace_err to log them.
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mabbas



Joined: 18 Nov 2009
Posts: 2

PostPosted: Thu Jan 21, 2010 8:17 pm    Post subject: Reply with quote

I had the same problem with WinSip but turned out to be issue on my end.

http://www.voip-info.org/boards/index.php?t=19148

Hope that helps.
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