Rob-san.
Thaks for your quick response.
The indication of SIPp is as follows.
A telegraphic message from SIPp does not seem to be recognized in the
Asterisk side.
As for the Asterisk side, an error and warning are displayed.
Only with a screen of this SIPp, do you understand a cause?
We having a lot of trouble.
======================= SIPp display ================================
[root@f1 sipp_packages]# sipp -sn uac -d 30000 -s 2005 192.168.1.101 -l 1
Warning: open file limit > FD_SETSIZE; limiting max. # of open files to
FD_SETSIZE = 1024
Resolving remote host '192.168.1.101'... Done.
------------------------------ Scenario Screen -------- [1-9]: Change
Screen --
Call-rate(length) Port Total-time Total-calls Remote-host
10.0(30000 ms)/1.000s 5060 16.34 s 163
192.168.1.101:5060(UDP)
3 new calls during 0.313 s period 1 ms scheduler resolution
0 calls (limit 1) Peak was 1 calls, after 0 s
0 Running, 166 Paused, 4 Woken up
0 dead call msg (discarded) 0 out-of-call msg (discarded)
3 open sockets
Messages Retrans Timeout
Unexpected-Msg
INVITE ----------> 163 0 0
100 <---------- 0 0 0 163
180 <---------- 0 0 0 0
183 <---------- 0 0 0 0
200 <---------- E-RTD1 0 0 0 0
ACK ----------> 0 0
Pause [ 30.0s] 0 0
BYE ----------> 0 0 0
200 <---------- 0 0 0 0
------------------------------ Test Terminated
--------------------------------
(2013/10/08 16:32), Rob Day wrote:
> On 8 October 2013 07:07, Yoshiaki Sato [Techno Business]
> <[email protected]> wrote:
>> I have begun to just use SIPp, but, in Asterisk1.8, an evaluation by UAC
>> is possible, but SIPp does not come in Invite in Asterisk11.03.
>
> Hi Yoshiaki,
>
> I'm not quite sure what you mean by "SIPp does not come in Invite in
> Asterisk11.03". If Asterisk 11.03 and 1.8 both expect to receive
> traffic from SIP clients, I would expect SIPp to work for both.
>
> If that doesn't answer your question, would you mind explaining the
> problem further?
>
> Best,
> Rob
>
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