Hi,
and if you try change "SIP/2.0 [field0] [field1]" with "SIP/2.0 486  
Busy" it still doesnt work? Or if you change your csv to:
=================================================
SEQUENTIAL
486;Busy here
================================================

Ondrej Cakan





Cituji sipprh <dados_rh-s...@yahoo.com.br>:

> Hello,
>
> Im trying to supply my script with the .csv file below, to test a gateway
> behavior when it receives SIP error codes, however I always get this  
> same error
> message from SIPp:
>
> 2010-10-05    10:41:27:696    1286286087.696511: Response codes must  
> be in the
> range of 100-700.
>
> ====SIPmessage.csv======
> SEQUENTIAL
> 486;Busy
> =====================
>
> Does anyone know how to fix this? 486 is still in range, so I dont know whats
> happening.
>
> My .xml script is shown below:
>
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>
> <!-- This program is free software; you can redistribute it and/or      -->
> <!-- modify it under the terms of the GNU General Public License as     -->
> <!-- published by the Free Software Foundation; either version 2 of the -->
> <!-- License, or (at your option) any later version.                    -->
> <!--                                                                    -->
> <!-- This program is distributed in the hope that it will be useful,    -->
> <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
> <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
> <!-- GNU General Public License for more details.                       -->
> <!--                                                                    -->
> <!-- You should have received a copy of the GNU General Public License  -->
> <!-- along with this program; if not, write to the                      -->
> <!-- Free Software Foundation, Inc.,                                    -->
> <!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
> <!--                                                                    -->
> <!--                 Sipp default 'uas' scenario.                       -->
> <!--                                                                    -->
>
> <scenario name="Basic UAS responder">
>   <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
>   <!-- are saved and used for following messages sent. Useful to test   -->
>   <!-- against stateful SIP proxies/B2BUAs.                             -->
> <recv request="INVITE" crlf="true">
> </recv>
>
>   <!-- The '[last_*]' keyword is replaced automatically by the          -->
>   <!-- specified header if it was present in the last message received  -->
>   <!-- (except if it was a retransmission). If the header was not       -->
>   <!-- present or if no message has been received, the '[last_*]'       -->
>   <!-- keyword is discarded, and all bytes until the end of the line    -->
>   <!-- are also discarded.                                              -->
>   <!--                                                                  -->
>   <!-- If the specified header was present several times in the         -->
>   <!-- message, all occurences are concatenated (CRLF seperated)        -->
>   <!-- to be used in place of the '[last_*]' keyword.                   -->
>
>    <send>
>     <![CDATA[
>
>       SIP/2.0 100 Trying
>       [last_Via:]
>       [last_From:]
>       [last_To:];tag=[pid]SIPpTag01[call_number]
>       [last_Call-ID:]
>       [last_CSeq:]
>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>       Content-Length: 0
>
>     ]]>
>   </send>
>
>   <send>
>     <![CDATA[
>
>       SIP/2.0 [field0] [field1]
>       [last_Via:]
>       [last_From:]
>       [last_To:];tag=[pid]SIPpTag01[call_number]
>       [last_Call-ID:]
>       [last_CSeq:]
>       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>       Content-Length: 0
>
>     ]]>
>   </send>
>
>
>   <recv request="ACK"
>         optional="true"
>         rtd="true"
>         crlf="true">
>   </recv>
>
>   <recv request="BYE">
>   </recv>
>
>
>
>   <!-- Keep the call open for a while in case the 200 is lost to be     -->
>   <!-- able to retransmit it if we receive the BYE again.               -->
>   <timewait milliseconds="4000"/>
>
>
>   <!-- definition of the response time repartition table (unit is ms)   -->
>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>
>   <!-- definition of the call length repartition table (unit is ms)     -->
>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>
> </scenario>
>
>
> </scenario>
>
> Files are attached.
>
> Thanks Advance,
> Renan
>
>
>




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