Hi,

I would design it like this:

<recv request="OPTIONS" optional="true" next="1">

</recv>

<recv request="INVITE"  next="2">

</recv>

<label id="1"/>
<!-- Sent 200 OK and jump to EOF(3) -->

<label id="2"/>
<!-- Process INVITE and jump after last msg to EOF(3)-->

<label id="3"/>
<!-- EOF -->

Niels


Quoting Jung Lim <[EMAIL PROTECTED]>:

> Hi,
>
> I want to use SIPp as simple UAS.
> The problem is that a sip proxy keeps on sending SIP OPTIONS to SIPp
> UAS for health check.
> For SIP OPTIONS message,  I want to send  send 200 OK.
> For SIP INVITE messgae, I want to send 100 Trying and process next
> requests as defined in the scenario xml file.
>
> Somehow, for SIP OPTIONS message, SIPp is sending 100 Trying...
>
> Can someone help what I am missing here?
>
>
>
>
>  <recv request="INVITE|OPTIONS" crlf="true" regexp_match="true" >
>     <action>
>        <ereg regexp="INVITE" search_in="hdr" start_line="true"
> assign_to="1" />
>        <ereg regexp="OPTIONS" search_in="hdr" start_line="true"
> assign_to="2" />
>     </action>
>  </recv>
>
>  <send test="1" next="1">
>    <![CDATA[
>
>      SIP/2.0 100 Trying
>      [last_Via:]
>      [last_From:]
>      [last_To:]
>      [last_CSeq:]
>      [last_Call-ID:]
>      Content-Length: 0
>
>    ]]>
>  </send>
>
>  <send retrans="500" next="2" test="2">
>    <![CDATA[
>
>      SIP/2.0 200 OK
>      [last_Via:]
>      [last_From:]
>      [last_To:];tag=[pid]SIPpTag01[call_number]
>      [last_Call-ID:]
>      [last_CSeq:]
>      Record-Route:[$3]
>      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>
>
> <label id="1"/>
>
>  <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:]
>      Record-Route:[$3]
>      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"
>        rtd="true"
>        crlf="true">
>  </recv>
>  <pause milliseconds="6000"/>
>  <send retrans="500">
>    <![CDATA[
>
>      BYE [$2] SIP/2.0
>      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>      From:[$4];tag=[pid]SIPpTag00[call_number]
>      To:[$5]
>      [last_Call-ID:]
>      CSeq: 3 BYE
>      Route:[$3]
>      Max-Forwards: 70
>      Content-Length: 0
>      Supported: timer
>
>    ]]>
>  </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"/>
> <label id="2"/>
> </scenario>
>
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>




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