[Sipp-users] why refuse to send ACK ..!

[hpvpl <va....@gm...>]

Hy, here is attached a simple script I would like to execute. This is a uac
script going through a proxy. I have an the other side (above the proxy) a
uas waiting for the message.

I've taken the uac.xml provided from the SIPP web site and updated with
ethereal raw dumpt from a scenario with real X-lite (and the proxy).

My problem is that the uac, on reception of the 200OK (of INVITE) never sen=
d
the ACK ... SIPP do not complains (no trace_err). How can I troubleshoot
such problem.
I'm a new user of SIPP. Perhaps the answer is trivial ..!
Thanks for your support

<?xml version=3D"1.0" encoding=3D"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 'uac' scenario. -->
<!-- -->

<scenario name=3D"Basic Sipstone UAC">
<send retrans=3D"500">
<![CDATA[

INVITE sip:si...@ii... SIP/2.0
Via: SIP/2.0/UDP 192.168.2.153:5060 <http://192.168.2.153:5060>
;rport;branch=3Dz9hG4bK0E2A20AC8E8D4EE3AD8506A6BBA2B452
From: sippuac <sip:si...@ii...>;tag=3D129464374
To: <sip:si...@ii...>
Contact: <sip:sippuac@192.168.2.153:5060>
Call-ID: 04480A22-1AB3-496E-A8E0-DD23FFE692F1@192.168.2.153
CSeq: 15061 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 296

v=3D0
o=3Dsippuac 7082283 7082283 IN IP4 192.168.2.153 <http://192.168.2.153>
s=3DX-Lite
c=3DIN IP4 192.168.2.153 <http://192.168.2.153>
t=3D0 0
m=3Daudio 8000 RTP/AVP 0 8 3 98 97 101
a=3Drtpmap:0 pcmu/8000
a=3Drtpmap:8 pcma/8000
a=3Drtpmap:3 gsm/8000
a=3Drtpmap:98 iLBC/8000
a=3Drtpmap:97 speex/8000
a=3Drtpmap:101 telephone-event/8000
a=3Dfmtp:101 0-15
]]>
</send>

<recv response=3D"100"
optional=3D"true">
</recv>

<recv response=3D"180" optional=3D"true">
</recv>

<recv response=3D"200" rtd=3D"true">
</recv>

<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost =3D "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[

ACK sip:sippuas@192.168.2.89:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.153:5060 <http://192.168.2.153:5060>
;rport;branch=3Dz9hG4bK0A57D4ADB33347F69C86E98279D1BBA7
From: sippuac <sip:si...@ii...>;tag=3D129464374
To: <sip:si...@ii...>;tag=3D3871825980
Contact: <sip:sippuac@192.168.2.153:5060>
Route: <sip:sippuas@192.168.2.84;ftag=3D129464374;lr=3Don>
Call-ID: 04480A22-1AB3-496E-A8E0-DD23FFE692F1@192.168.2.153
CSeq: 15061 ACK
Max-Forwards: 70
Content-Length: 0

]]>
</send>

<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value=3D"10, 20, 30, 40, 50, 100, 150, 200"/>

<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value=3D"10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>