Could you send your exact xml scenario? And could you possibly make a network 
capture (using Ethereal/Wireshark/tcpdump)?

I'm sending my test scenario in attachment. I don't know if the sipp-users 
list accepts attachment, but I'm also sending CC to your e-mail. My test 
scenario reads record-route from INVITE and puts it, after splittion, to 200 
OK. I don't know what you exactly want to do. It seems you are putting it 
from one invite to another. It would be more clear if you send your xml file.

Kind regards
-- 
Tomasz Radziszewski
Senior Software Engineer
Ericpol Telecom sp. z o.o.
Madalinskiego 9, 30-303 Krakow, Poland
e-mail: [EMAIL PROTECTED]
tel: +48 (12) 295 48 16
mobile: +48 663 426 706
http://www.ericpol.pl/

> hey
>
> I tried with the following format u have suggested still then it it not
> working and its sending Route header as blank for hold invite can try it on
> u r m/c because it not working from my part.
>
> can u have a look at the regexp again and i am also using same version
> 1.1rc6
>
> On Fri, Apr 25, 2008 at 12:30 PM, Tomasz Radziszewski
>
> <[EMAIL PROTECTED]> wrote:
> > Hi
> >
> > This seems to work for me, with SIPp 1.1rc6 (description below)
> > ----------------------
> >  <recv request="INVITE" crlf="true">
> >    <action>
> >      <ereg
> > regexp="([^,]*),([^,]*),([^,]*),([^,]*),([^,]*),([^,]*),([^,]*)"
> > search_in="hdr" header="Record-Route: " assign_to="1,2,3,4,5,6,7,8"/>
> > </action>
> >  </recv>
> >
> >  <send retrans="500">
> >    <![CDATA[
> >
> >      SIP/2.0 200 OK
> >      [last_Via:]
> >      [last_From:]
> >      [last_To:];tag=[call_number]
> >      [last_Call-ID:]
> >      [last_CSeq:]
> >      Record-Route: [$8]
> >      Record-Route: [$7]
> >      Record-Route: [$6]
> >      Record-Route: [$5]
> >      Record-Route: [$4]
> >      Record-Route: [$3]
> >      Record-Route: [$2]
> >      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>
> > -------------------
> > When receiving INVITE, we split the Record-Route header into several
> > subexpressions, using parentheses () and assign them to variables
> > 1,2,3,4,5,6,7,8.
> >
> > Variable 1 gets the whole expression, because SIPp works this way, and we
> > ignore it. Variables 2-8 get all the parts. Each part is [^,]* which
> > means: any character other than comma, repeated any number of times.
> >
> > Using record-route from your example:
> > <sip:10.106.5.219:5060;lr>;term,<sip:[EMAIL PROTECTED]:5060
> > ;lr>;term,<sip:[EMAIL PROTECTED]:5060;lr>...
> >
> > variable 2 gets value: <sip:10.106.5.219:5060;lr>;term
> > variable 3: <sip:[EMAIL PROTECTED]:5060;lr>;term
> > etc.
> >
> > Then we use the variables 2-8 in reverse order to construct the
> > record-route
> > in 200 OK (the expressions [$8] etc)
> >
> > Note that we assume that your Record-Route always has 7 parts. It would
> > be more complex (impossible?) to make a regular expression accepting any
> > number
> > of parts.
> >
> > I've got this output, using this regular expression and the Record-route
> > from
> > your example:
> >
> > Record-Route: <sip:10.106.5.219:5060;lr>;orig
> > Record-Route: <sip:[EMAIL PROTECTED]:5060;lr>
> > Record-Route: <sip:[EMAIL PROTECTED]:5060;lr>
> > Record-Route: <sip:[EMAIL PROTECTED]:5060;lr>;term
> > Record-Route: <sip:[EMAIL PROTECTED]:5060;lr>
> > Record-Route: <sip:[EMAIL PROTECTED]:5060;lr>;term
> > Record-Route: <sip:10.106.5.219:5060;lr>;term
> >
> > I'm not sure if you want the words "Record-route" on each line, but you
> > can simply remove them from the lines "Record-Route: [$8]" etc. Similarly
> > you can
> > add comma at the end of each line if you want it (as in your post).
> >
> > Best regards
> > --
> > Tomasz Radziszewski
> > Senior Software Engineer
> > Ericpol Telecom sp. z o.o.
> > Madalinskiego 9, 30-303 Krakow, Poland
> > e-mail: [EMAIL PROTECTED]
> > tel: +48 (12) 295 48 16
> > mobile: +48 663 426 706
> > http://www.ericpol.pl/
> >
> > > HI
> > >
> > > I am running a scenario where A calling user B and after media path is
> > > established. B is putting A on hold where c=0.0.0.0.
> > >
> > > I need to extract Record route from the Invite that was recieved from
> > > the Invite by B. This can be used by using Regexp. i have tried with
> > > diffrent combination.when i tried with a regexp that works in regexp
> > > calculator
> >
> > but
> >
> > > its not working in the xml script because sipp parser is unable to
> > > parse the script.
> > >
> > > This is what i am getting in the Invite
> > >
> > > Record-Route: <sip:10.106.5.219:5060
> > > ;lr>;term,<sip:[EMAIL PROTECTED]:5060
> > > ;lr>;term,<sip:[EMAIL PROTECTED]:5060
> > > ;lr>,<sip:[EMAIL PROTECTED]:5060
> > > ;lr>;term,<sip:[EMAIL PROTECTED]:5060
> > > ;lr>,<sip:[EMAIL PROTECTED]:5060;lr>,<sip:10.106.5.219:506
> > >0 ;lr>;orig
> > >
> > >  when i want to send in the invite it should get reversed and all the
> > > record route should be in diffrent line.
> > >
> > > <sip:10.106.5.219:5060;lr>;orig
> > > <sip:[EMAIL PROTECTED]:5060;lr>,
> > > <sip:[EMAIL PROTECTED]:5060;lr>,
> > > <sip:[EMAIL PROTECTED]:5060;lr>;term,
> > > <sip:[EMAIL PROTECTED]:5060;lr>,
> > > <sip:[EMAIL PROTECTED]:5060;lr>;term,
> > > <sip:10.106.5.219:5060;lr>;term,
> > >
> > >
> > > this is the format what i need to send in Invite because it is the
> > > format what the soft phone sends
> > >
> > > until i send in this format , this message is not getting routed and i
> > > am getting 500 error message "unable to process Invite"
> > >
> > > So need help regarding this!!!!!!!!!!!!
> > >
> > >
> > > Its very Urgent
> >
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<?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 'uas' scenario.                       -->
<!--                                                                    -->

<scenario name="Basic UAS responder">
  <!-- 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 request="INVITE" crlf="true">
    <action>
      <ereg regexp="([^,]*),([^,]*),([^,]*),([^,]*),([^,]*),([^,]*),([^,]*)" search_in="hdr" header="Record-Route: " assign_to="1,2,3,4,5,6,7,8"/>
    </action>
  </recv>

  <send retrans="500">
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Record-Route: [$8]
      Record-Route: [$7]
      Record-Route: [$6]
      Record-Route: [$5]
      Record-Route: [$4]
      Record-Route: [$3]
      Record-Route: [$2]
      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.               -->
  <pause 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>

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