Hi,

I am trying to establish call through a inf file with different IP address
through the command line as described in the reference document
./sipp -sf client_ipphone1.xml -t ui -inf data1.txt -ip_field 2 <my local
machine ip address>

but I am getting a error "Unable to bind main socket, errno = 99 (Cannot
assign requested address)"

Attached is my data1.txt and scenario file.

-- 
Thank you  with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
SEQUENTIAL
Gopal;2001;192.168.0.64
Murali;1777;203.196.148.210
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!--./sipp -i 192.168.0.83 -p 6060 -sf client1.xml -d 20000 192.168.0.58 -trace_err -->

<scenario name="UAClient">
<send retrans="500">
      <![CDATA[
  
        INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
        Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
		From: [field0] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
        To: sut <sip:[service]@[remote_ip]:[remote_port]>
        Call-ID: [call_id]
        CSeq: 1 INVITE
        Contact: sip:[field1]@[local_ip]:[local_port]
        Max-Forwards: 70
        Subject: Performance Test
        Content-Type: application/sdp
        Content-Length: [len]
  
        v=0
        o=user1 53655765 2353687637 IN IP4 127.0.0.1
        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 response="100"
          optional="true">
    </recv>
  
    <recv response="180" 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">
    </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:[service]@[remote_ip]:[remote_port] SIP/2.0
        Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
        From: sipp <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
        To: sut <sip:[service]@[remote_ip]:[remote_port]>
		Call-ID: [call_id]
        CSeq: 1 ACK
        Contact: sip:[field1]@[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:[field1]@[remote_ip]:[remote_port] SIP/2.0
        Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
        From: sipp <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
		To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
        Call-ID: [call_id]
        CSeq: 2 BYE
        Contact: sip:[field1]@[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"/>
   <pause milliseconds="2000"/>
</scenario>
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