Hi.
When I run sipp, I get the following error: Authentication keyword without
dialog_authentication!. and sipp doesn't send any message to asterisk.
See attached xml and csv files.
sipp -v output:
SIPp v3.1-TLS-PCAP, version svn452, built Jul 4 2008, 13:21:28.
Thank you.
SEQUENTIAL,PRINTF=6
Cabin %01d;19_cabin_%01d;[authentication username=19_cabin_%01d password=19_cabin_%01d];0044%01d018084%01d;
<?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="Asterisk/OpenPBX Agent login - responder">
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
From: [field1] <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number]
To: [field1] <sip:[fiel...@[local_ip]:[local_port]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Content-Length: 0
Expires: 300
[field2]
]]>
</send>
<!-- simple case - just jump over a line -->
<recv response="200" rtd="true" next="5">
</recv>
<recv response="200">
</recv>
<label id="5"/>
<pause milliseconds="2000" crlf="true" />
<!-- Now we need to log the agent on! -->
<!-- For that we need to dial the agent route -->
<!-- So let's start with an invite -->
<send retrans="500">
<![CDATA[
INVITE sip:[fiel...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field1] <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number]
To: sip:[fiel...@[remote_ip]:[remote_port]
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
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
a=rtpmap:3 GSM/8000
]]>
</send>
<recv response="407" auth="true" next="3">
</recv>
<recv response="401" auth="true" next="3">
</recv>
<label id="3"/>
<!-- We have been challenged, lets show them the money! And invite again -->
<!-- Send the acnoledgement for the 407 message -->
<send>
<![CDATA[
ACK sip:[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: [field1] <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number]
To: sip:[fiel...@[remote_ip]:[remote_port]
[last_Call-ID]
[last_CSeq]
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[fiel...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field1] <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number]
To: sip:[fiel...@[remote_ip]:[remote_port]
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
[field2]
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
a=rtpmap:3 GSM/8000
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<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:[field1]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: [field1] <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number]
To: sip:[fiel...@[remote_ip]:[remote_port]
[last_Call-ID]
[last_CSeq]
Contact: <sip:[fiel...@[local_ip]:[local_port]>
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:[fiel...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[fiel...@[field1]>
To: sut <sip:[fiel...@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:s...@[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"/>
</scenario>
------------------------------------------------------------------------------
SOLARIS 10 is the OS for Data Centers - provides features such as DTrace,
Predictive Self Healing and Award Winning ZFS. Get Solaris 10 NOW
http://p.sf.net/sfu/solaris-dev2dev
_______________________________________________
Sipp-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/sipp-users
