Try Following, It should work for user case

command -> sipp -sn uac <server ip> -sf scenario.xml -m 1

I registered user 2000 to registrar with no password, in your case change 
scenario accordingly .

<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="UAC Register">
 
    <send retrans="500">
        <![CDATA[
 
        REGISTER sip:2000@[remote_ip]:[remote_port] SIP/2.0
        Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
        From: v4-in 
<sip:v4-in@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
        To: <sip:v4-in@[remote_ip]:[remote_port]>
        Call-ID: [call_id]
        CSeq: 1 REGISTER
        Contact: sip:v4-in@[local_ip]:[local_port]
        Max-Forwards: 70
        Subject: REGISTER Test
        Expires: 3600
        Content-Length: 0
 
        ]]>
    </send>
 
    <recv response="200" rtd="true" />

  <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/2.0 180 Ringing
      [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 retrans="500">
    <![CDATA[

      SIP/2.0 200 OK
      [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-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

  <recv request="ACK"
        optional="true"
        rtd="true"
        crlf="true">
  </recv>

  <recv request="BYE">
  </recv>

  <send>
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>

  <!-- 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>


Best Regards,Sakharam Thorat.

Date: Fri, 10 Jan 2014 13:37:34 -0500
From: sangdr...@gmail.com
To: sipp-users@lists.sourceforge.net
Subject: [Sipp-users] UAC receive INVITE problems

I am trying to have a UAC xml script that can log into my server, then recieve 
an INVITE and respond to it.  Currently I can register, and even place calls 
just fine.  I seem to be having issues with the recieving of the invite.  Sipp 
see's the invite and says it can't be mapped to a known sipp call, even though 
my scenario is sitting and waiting for an INVITE.  Can someone help me 
understand how to use the recv?

Currently I do the following, and it never considers the INVITE as part of my 
call so I get stuck

<scenario name="User Register"><send>

    <![CDATA[        REGISTER sip:[remote_ip] SIP/2.0        Via: 
SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]        From: 
<sip:[field0]@[field1]>;tag=[call_number]
        To: <sip:[field0]@[field1]>        Call-ID: [call_id]        CSeq: 
[cseq] REGISTER        Contact: 
<sip:[field0]@[local_ip]:[local_port];transport=tls>        Max-Forwards: 10
        Expires: 120        User-Agent: SIPp        Content-Length: 0    
]]></send><recv response="401" auth="true"></recv>
<send>    <![CDATA[        REGISTER sip:[remote_ip] SIP/2.0        Via: 
SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]        From: 
<sip:[field0]@[field1]>;tag=[call_number]
        To: <sip:[field0]@[field1]>        Call-ID: [call_id]        CSeq: 
[cseq] REGISTER        Contact: 
<sip:[field0]@[local_ip]:[local_port];transport=tls>        [field2]
        Max-Forwards: 10        Expires: 120        User-Agent: SIPp        
Content-Length: 0    ]]></send><recv response="200"></recv>

<recv request="INVITE"></recv>


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