Hi,

Thanks for prompt response!

Actual I want to make the calls to my PBX (UAS). I try to register the one user 
it working fine but I unable to make the calls to UAS.

Please find my attached inject files and command with error logs .


root@hss-virtual-machine:/opt/sipp# ./sipp 10.112.87.177 -sf INVITE_client.xml 
-i 10.112.87.132 -inf INVITE_client.csv -m 1 -l 1 -trace_msg -trace_err
2016-11-07      22:45:29:962    1478538929.962488: No name specified for 'file' 
keyword!
.
root@hss-virtual-machine:/opt/sipp#


10.112.87.177 is my UAS (PBX) and 10.112.87.132 local ip(Client).


Regards
Surender Singh



From: Pietro Bertera [mailto:pie...@bertera.it]
Sent: 07 November 2016 16:24
To: Surender Singh <surende...@hcl.com>
Cc: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] Customize XML Scenarios

Hi,

you can use customised call scenarios using the -sf option.

How to create your own scenario is well documented here: 
http://sipp.sourceforge.net/doc3.3/reference.html#Create+your+own+XML+scenarios

or you can start from one one the builtin scenarios: sipp -sd uas will dump the 
uas builtin scenario.

From your email isn't clear what you are trying to achieve, I hope this info 
will help.

Thanks,

Hello,

I am new for SIPp and want use the customized XML for call scenarios.

Can I us the sipp for Supplementary Services automation (call forwarding, 
transfer ,three part etc.).

Actual I want to automate these services  .Please suggest me whether this is 
suitable.



--
Bertera Pietro
http://www.bertera.it


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Attachment: REGISTER_client.csv
Description: REGISTER_client.csv

<?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 'uac' scenario.                       -->
<!--                                                                    -->

<scenario name="Basic Sipstone UAC">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
  <send retrans="500">
    <![CDATA[
        INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
	Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
	From: <sip:[field1]@[remote_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
	To: <sip:[filed0]@[remote_ip]:[remote_port]>
	Call-ID: [call_id]
	CSeq: 1 INVITE
	Contact: sip:[field1]@[local_ip]:[local_port]
	Max-Forwards: 70
	Subject: Basic call
	Content-Type: application/sdp
	Content-Length: [len]
	v=0
	o=Z 0 0 IN IP4 10.112.94.241
	s=Z
	c=IN IP4 10.112.94.241
	t=0 0
	m=audio 8000 RTP/AVP 3 110 8 0 97 101
	a=rtpmap:110 speex/8000
	a=rtpmap:97 iLBC/8000
	a=fmtp:97 mode=30
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=sendrecv
    ]]>
  </send>

  <recv response="100"
        optional="true">
  </recv>

  <recv response="180" optional="true">
  </recv>

  <recv response="183" optional="true">
  </recv>

  <!-- 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 response="200" rtd="true" rrs="true">
  </recv>

  <!-- Packet lost can be simulated in any send/recv message by         -->
  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  <send>
    <![CDATA[
      ACK sip:[field1]@[filed2]:[remote_port] SIP/2.0
      [routes]
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[filed0]@[remote_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      To: <sip:[filed1]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:[]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0
    ]]>
  </send>

  <!-- This delay can be customized by the -d command-line option       -->
  <!-- or by adding a 'milliseconds = "value"' option here.             -->
  <pause/>

  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  <send retrans="500">
    <![CDATA[
      BYE sip:[field0]@[field2]:[remote_port] SIP/2.0
      [routes]
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[remote_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 BYE
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0
    ]]>
  </send>

  <recv response="200" crlf="true">
  </recv>

  <!-- 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>
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