Hello Ondrej,
Thanks for the quick reply!
Ive just tried both approaches you've suggested:
1 - If I use "SIP/2.0 486 Busy"
- It Works. The problem only happens when SIPp reads from .csv file
- But, since I'd like to test several "SIP Error responses",instead of creating
several scripts for each message, Id like to use only one script capable of
reading several "SIP Error responses from a .csv file. e.g below:
==================file.csv======================
SEQUENTIAL
486;Busy here
606;Not Acceptable
603;Decline
..and so on....
================================================
2 - Ive also tested changing "Busy" to "Busy Here" and got the same error.
3 - And is also strange, since if add [field0] to another part of the SIP
message like:
<sip:[field0]:[local_ip]:[local_port];transport=[transport]>
SIPp replaces [field0] with the number from .csv file perfectly!
So, the problem only happens when I try to replace specifically the line
"SIP/2.0 [field0] Busy here" with a number from .csv file.
Thanks again!Im still trying other approaches here.
Best Regards,
Renan
________________________________
De: "cakan...@fel.cvut.cz" <cakan...@fel.cvut.cz>
Para: sipprh <dados_rh-s...@yahoo.com.br>
Cc: "sipp-users@lists.sourceforge.net" <sipp-users@lists.sourceforge.net>
Enviadas: Terça-feira, 5 de Outubro de 2010 13:42:31
Assunto: Re: [Sipp-users] Res: Sipp-users Digest, Vol 53, Issue 1
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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