uas-hold.xml

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="Basic UAS responder">
  <recv request="INVITE" crlf="true">
   <action>
    <!-- since we need to send a request to the remote part -->
    <!-- we need to extract the Contact and the From header content -->
    <ereg regexp=".*" search_in="hdr" header="From" assign_to="remote_from"/>
    <!-- assign the content of the Contaact SIP URI to the remote_contact var -->
    <!-- first var of assign_to contains the whole match -->
    <ereg regexp="sip:(.*)>.*" search_in="hdr" header="Contact" assign_to="trash,remote_contact"/>
   </action>
  </recv>

  <!-- since SIPp complains about not used variable reference the trach var -->
  <Reference variables="trash"/>  

  <send>
    <![CDATA[

      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[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=[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=sipp 87308505 1 IN IP[local_ip_type] [local_ip]
      s=-
      t=0 0
      m=audio [media_port] RTP/AVP 8 101
      c=IN IP[media_ip_type] [media_ip]
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-15
      a=ptime:20

    ]]>
  </send>

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

  <pause milliseconds="3000" crlf="true"/>

  <nop display=">>>>> Sending hold re-INVITE <<<<" crlf="true"/>
  
  <send retrans="500">
   <![CDATA[
      INVITE [$remote_contact] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: sipp  <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: [$remote_from]
      [last_Call-ID:]
      CSeq: 1 INVITE
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=sipp 87308505 2 IN IP[local_ip_type] [local_ip]
      s=-
      t=0 0
      m=audio [media_port] RTP/AVP 8 101
      c=IN IP[media_ip_type] [media_ip]
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-15
      a=ptime:20
      a=sendonly
    ]]>
   </send> 
 
   <recv response="100" optional="true">
   </recv>

   <recv response="200">
   </recv> 
   
   <send>
    <![CDATA[ 
     ACK [$remote_contact] SIP/2.0
     Via: SIP/2.0/[transport] [local_ip]:[local_port]
     From: sipp  <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
     To: [$remote_from]
     [last_Call-ID:]
     CSeq: 1 ACK
     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
    ]]>
   </send>

  <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.               -->
  <pause milliseconds="2000"/>

</scenario>