Hi all,
       i just discovered my problem, i have sipp v3.1 on my kubuntu machine, 
and also i have installed (sip-tester) package which is one of the packages of 
kubuntu and it is actually sipp v2.0, so when i type the command:
 
sipp -sn uas ( it loads v 2.0)
 
when i want to load v3.1 i have to do that under the installation 
directory.....but now i removed that package v2.0 to make no conflict.
 
thanks all
 
Amar



> From: [EMAIL PROTECTED]> Subject: Sipp-users Digest, Vol 29, Issue 49> To: 
> [email protected]> Date: Fri, 24 Oct 2008 10:02:47 +0000> > 
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> replying, please edit your Subject line so it is more specific> than "Re: 
> Contents of Sipp-users digest..."> > > Today's Topics:> > 1. Log action with 
> time (Le Van Tu)> 2. Fwd: local_port (raul derby)> 3. Re: inserting timestamp 
> of messages (amar mahmoud)> 4. i can not listen the voice in softphone 
> (jordan pan)> > > 
> ----------------------------------------------------------------------> > 
> Message: 1> Date: Thu, 23 Oct 2008 16:33:02 -0400> From: "Le Van Tu" <[EMAIL 
> PROTECTED]>> Subject: [Sipp-users] Log action with time> To: 
> [email protected]> Message-ID:> <[EMAIL PROTECTED]>> 
> Content-Type: text/plain; charset="iso-8859-1"> > Hi folks,> > I must use log 
> action with -trace_log. I wonder if there's way to log> something with 
> time-stamp ?> > For example> > <nop>> <action>> <log message="Receive call 
> from something for users [$10] *in time :> ??????*"/>> </action>> </nop>> > 
> Regards,> > Tu> -------------- next part --------------> An HTML attachment 
> was scrubbed...> > ------------------------------> > Message: 2> Date: Thu, 
> 23 Oct 2008 15:57:22 -0500> From: "raul derby" <[EMAIL PROTECTED]>> Subject: 
> [Sipp-users] Fwd: local_port> To: [email protected]> 
> Message-ID:> <[EMAIL PROTECTED]>> Content-Type: text/plain; 
> charset="iso-8859-1"> > ---------- Forwarded message ----------> From: raul 
> derby <[EMAIL PROTECTED]>> Date: Tue, Oct 21, 2008 at 9:58 AM> Subject: 
> local_port> To: [email protected]> > > Hi> > The [local_port] 
> field is using port 5060 as the default port which varies> from the source 
> port which is dynamically generated from the TCP/IP stack.> > I earnestly 
> request you to let me know the sipp option where the local_port> value = 
> source port value dynamically generated from the TCP/IP stack.> > Kind 
> regards,> > RAUL> -------------- next part --------------> An HTML attachment 
> was scrubbed...> > ------------------------------> > Message: 3> Date: Thu, 
> 23 Oct 2008 22:06:19 +0000> From: amar mahmoud <[EMAIL PROTECTED]>> Subject: 
> Re: [Sipp-users] inserting timestamp of messages> To: Charles P Wright 
> <[EMAIL PROTECTED]>> Cc: [EMAIL PROTECTED],> 
> "[email protected]" <[email protected]>> 
> Message-ID: <[EMAIL PROTECTED]>> Content-Type: text/plain; 
> charset="iso-8859-1"> > > Hi Charles and all,> I have used [timestamp] many 
> times and it was working, just now it is not working for me , i tried it in 3 
> PCs, when i run a scenario which has [timestamp] ; it gives the error:> 
> Unsupported keyword 'timestamp' in xml scenario file.> > really strange, i 
> even run the same file which was working before and now it gives me that 
> error despite i m using the same version 3.1.> > anyone has advice.> > Amar> 
> ----------------------------------------> > To: [EMAIL PROTECTED]> > CC: 
> [email protected]; [EMAIL PROTECTED]> > Subject: Re: 
> [Sipp-users] inserting timestamp of messages> > From: [EMAIL PROTECTED]> > 
> Date: Mon, 15 Sep 2008 13:56:05 -0400> > > > You can use -trace_logs and the 
> [timestamp] keyword inside of a message="[timestamp]" /> action.> > > > 
> Charles> > > > [EMAIL PROTECTED] wrote on 09/09/2008 09:01:44 PM:> > > >> 
> Hi,> >> when using trace_msg options I can get the message with its > >> 
> timestamp, But I need to extract only the timestamp in a CVS file > >> .... i 
> have tried to execute external command through but > >> got problem with 
> that. > >> > >> Anyone who can help with that.> >> > >> thanks,> >> > >> Amar 
> Ahmed> >> > >> Want to do more with Windows Live? Learn ?10 hidden secrets? 
> from Jamie. > > > >> Learn Now> >> > > 
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>  > > ------------------------------> > Message: 4> Date: Fri, 24 Oct 2008 
> 17:58:23 +0800> From: "jordan pan" <[EMAIL PROTECTED]>> Subject: [Sipp-users] 
> i can not listen the voice in softphone> To: 
> [email protected]> Message-ID:> <[EMAIL PROTECTED]>> 
> Content-Type: text/plain; charset="iso-8859-1"> > Hi everyone,> > when i use 
> the sipp send a call to softphone, i can not listen the voice> in phone.I 
> have try many times,and feel very terrible. anyone can help me.> > Thanks in 
> advance!> > The following is xml files:and the pcap files in attachment.> > 
> <?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 'uac' scenario with pcap (rtp) play -->> <!-- -->> <scenario 
> name="UAC with media">> <!-- In client mode (sipp placing calls), the Call-ID 
> MUST be -->> <!-- generated by sipp. To do so, use [call_id] keyword.> -->> 
> <send retrans="500">> <![CDATA[> INVITE sip:[EMAIL PROTECTED]:[remote_port] 
> SIP/2.0> Via: SIP/2.0/[transport]> [local_ip]:[local_port];branch=[branch]> 
> From: [field0]> <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]> To: 
> [field1] <sip:[EMAIL PROTECTED]:[remote_port]>> Call-ID: [call_id]> CSeq: 1 
> INVITE> Contact: sip:[EMAIL PROTECTED]:[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[local_ip_type] [local_ip]> t=0 0> m=audio [auto_media_port] RTP/AVP 
> 8> a=rtpmap:8 PCMU/8000> a=rtpmap:101 telephone-event/8000> a=fmtp:101 
> 0-11,16> ]]>> </send>> <recv response="407" auth="true">> </recv>> <!-- 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. -->> <!-- 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:[EMAIL PROTECTED]:[remote_port] 
> SIP/2.0> Via: SIP/2.0/[transport]> [local_ip]:[local_port];branch=[branch]> 
> From: [field0]> <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]> To: 
> [field1]> <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]> Call-ID: 
> [call_id]> CSeq: 1 ACK> Contact: sip:[EMAIL PROTECTED]:[local_port]> 
> Max-Forwards: 70> Subject: Performance Test> Content-Length: 0> ]]>> </send>> 
> > <send retrans="500">> <![CDATA[> INVITE sip:[EMAIL PROTECTED]:[remote_port] 
> SIP/2.0> Via: SIP/2.0/[transport] [local_ip]:[local_port]> From: [field0]> 
> <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]> To: [field1] 
> <sip:[EMAIL PROTECTED]:[remote_port]>> Call-ID: [call_id]> CSeq: 2 INVITE> 
> Contact: sip:[EMAIL PROTECTED]:[local_port]> [field2]> 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=-> 
> t=0 0> c=IN IP[media_ip_type] [media_ip]> m=audio [auto_media_port] RTP/AVP 
> 8> a=rtpmap:8 PCMU/8000> ]]>> </send>> > <recv response="100" 
> optional="true"/>> <recv response="180" optional="true"/>> <recv 
> response="200" rtd="true" crlf="true"/>> > <send >> <![CDATA[> ACK sip:[EMAIL 
> PROTECTED]:[remote_port] SIP/2.0> Via: SIP/2.0/[transport]> 
> [local_ip]:[local_port];branch=[branch]> From: [field0]> <sip:[EMAIL 
> PROTECTED]:[local_port]>;tag=[call_number]> To: [field1]> <sip:[EMAIL 
> PROTECTED]:[remote_port]>[peer_tag_param]> Call-ID: [call_id]> CSeq: 2 ACK> 
> Contact: sip:[EMAIL PROTECTED]:[local_port]> Max-Forwards: 70> Subject: 
> Performance Test> Content-Length: 0> ]]>> </send>> > <!-- Play an pre-recorde 
> PCAP file (RTP stream) for testing tone -->> > <pause milliseconds="60000"/>> 
> <nop>> <action>> <exec play_pcap_audio="pcap/g711u/g711u_01.pcap"/>> 
> </action>> </nop>> <pause milliseconds="60000"/>> > <pause 
> milliseconds="60000"/>> > <!-- The 'crlf' option inserts a blank line in the 
> statistics report. -->> > <send retrans="500">> <![CDATA[> BYE sip:[EMAIL 
> PROTECTED]:[remote_port] SIP/2.0> Via: SIP/2.0/[transport] 
> [local_ip]:[local_port];branch=[branch]> From: [field0]> <sip:[EMAIL 
> PROTECTED]:[local_port]>;tag=[call_number]> To: [field1] <sip:[EMAIL 
> PROTECTED]:[remote_port]>[peer_tag_param]> Call-ID: [call_id]> CSeq: 2 BYE> 
> Contact: sip:[EMAIL PROTECTED]:[local_port]> Max-Forwards: 70> Subject: 
> Performance Test> Content-Length: 0> ]]>> </send>> <recv response="200" 
> crlf="true">> </recv>> > <send>> <![CDATA[> ACK sip:[EMAIL 
> PROTECTED]:[remote_port] SIP/2.0> Via: SIP/2.0/[transport]> 
> [local_ip]:[local_port];branch=[branch]> From: [field0]> <sip:[EMAIL 
> PROTECTED]:[local_port]>;tag=[call_number]> To: [field1]> <sip:[EMAIL 
> PROTECTED]:[remote_port]>[peer_tag_param]> Call-ID: [call_id]> CSeq: 2 ACK> 
> Contact: sip:[EMAIL PROTECTED]:[local_port]> Max-Forwards: 70> Subject: 
> Performance Test> Content-Length: 0> ]]>> </send>> > <pause 
> milliseconds="2000"/>> > > <!-- 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>> -- > Best regards!> jordan pan> Location:Shenzhen 
> China Company:www.justcall.cn> -------------- next part --------------> An 
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> Sipp-users Digest, Vol 29, Issue 49> 
> ******************************************
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