This is the script which sends the PCAP file to our IVR platform - is my media
type correct? Is the codec correct? Should I hear any audio to the called
party? I am trying to determine why I would not hear audio to the called party
playing the PCAP file. Please let me know if I am asking the wrong questions.
I have sent several emails to the list and no one is answering.
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=SIP Media Capabilities
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 18 0 2 96
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap96 telephone-event/8000
a=fmtp:96 0-15
a=fmtp:18 annexb=no
a=fmtp:96 0-15
a=maxptime:20
a=sendrecv]]>
</send>
<recv response="100" crlf="true" />
<recv response="183" crlf="true" />
<send>
<![CDATA[
PRACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 PRACK
Max-Forwards: 70
RAck: 1 1 INVITE
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true" />
<recv response="200" crlf="true" />
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<nop>
<action>
<exec play_pcap_audio="pcap\MMdemoMainmenuAMR.pcap"/>
</action>
</nop>
<pause milliseconds="360000" />
<send>
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 BYE
Max-Forwards: 70
Content-Length: 0
-----Original Message-----
From: sipp-users-requ...@lists.sourceforge.net
[mailto:sipp-users-requ...@lists.sourceforge.net]
Sent: Tuesday, February 22, 2011 8:14 AM
To: sipp-users@lists.sourceforge.net
Subject: Sipp-users Digest, Vol 57, Issue 9
Send Sipp-users mailing list submissions to
sipp-users@lists.sourceforge.net
To subscribe or unsubscribe via the World Wide Web, visit
https://lists.sourceforge.net/lists/listinfo/sipp-users
or, via email, send a message with subject or body 'help' to
sipp-users-requ...@lists.sourceforge.net
You can reach the person managing the list at
sipp-users-ow...@lists.sourceforge.net
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Sipp-users digest..."
Today's Topics:
1. Re: DTMF pcap files with payload type=127 (Michael Hirschbichler)
2. Re: Fwd: Beginning with SIPp - How to run scenarios
(Jos? Pablo M?ndez Soto)
3. Timeout attribute on initial recv request (John McNamara)
4. Discarded message which can't be mapped & Can't Hear Audio to
called party. (Corcione, John)
----------------------------------------------------------------------
Message: 1
Date: Mon, 21 Feb 2011 12:39:36 +0100
From: Michael Hirschbichler <s...@hirschbichler.biz>
Subject: Re: [Sipp-users] DTMF pcap files with payload type=127
To: sipp-users@lists.sourceforge.net
Message-ID: <4d624ef8.2090...@hirschbichler.biz>
Content-Type: text/plain; charset=ISO-8859-1
Hi,
untested: create a SIPp-(UAC)Scenario with a SDP containing only the a
common Voice-pt (like 0 or 8) and the pt=127 as you want. Then start
Wireshark, run the scenario and call a SIP Hardphone directly. On this
hardphone push the buttons in the wished order. In your wireshark-Trace,
you should now have the DTMF,
hth
br
Michael
On 2011-02-18 06:23, Shah, Nisha N. (Nisha) wrote:
> Hello,
>
> Does anyone know how I can generate DTMF digit pcap files with payload
> type =127? The digit pcap files that I downloaded from the SIPp
> website have a payload type=101, and currently I need to test with the
> 127 payload type.
>
> Any help is appreciated.
>
> Thanks,
> Nisha
>
>
>
> ------------------------------------------------------------------------------
> The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE:
> Pinpoint memory and threading errors before they happen.
> Find and fix more than 250 security defects in the development cycle.
> Locate bottlenecks in serial and parallel code that limit performance.
> http://p.sf.net/sfu/intel-dev2devfeb
>
>
>
> _______________________________________________
> Sipp-users mailing list
> Sipp-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/sipp-users
------------------------------
Message: 2
Date: Mon, 21 Feb 2011 06:32:56 -0600
From: Jos? Pablo M?ndez Soto <aux...@gmail.com>
Subject: Re: [Sipp-users] Fwd: Beginning with SIPp - How to run
scenarios
To: mayamatakeshi <mayamatake...@gmail.com>,
sipp-users@lists.sourceforge.net
Message-ID:
<AANLkTi=_2M=dnyhx1ncdgn0v_y4pc1vrswdm0n8d5...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Thank you mayamatakeshi,
appreciate your attention.
*JPMS
*********
2011/2/21 mayamatakeshi <mayamatake...@gmail.com>
>
> 2011/2/21 Jos? Pablo M?ndez Soto <aux...@gmail.com>
>
> Hello,
>>
>> I am completely lost right now trying to run SIPp, at least got version
>> 3.2 to compile with <make playpcap_ossl> option, but I have no idea how to
>> run it against my Asterisk box with the simplest scenario.
>>
>
> Read:
> http://sipp.sourceforge.net/doc3.0/reference.html
>
> To just send 1 single call to Asterisk you could use something like this:
> sipp -s DestNumber -i YourLocalIp -p YourLocalPort -m 1 -sn uac
> AsteriskAddress
>
>
>>
>> Other 2 questions I have:
>>
>> 1. Is the frontend for Windows only?
>>
>>
> No idea. I don't use Windows.
>
>>
>> 1. What am I missing when getting the following (trying to dump a
>> default scenario):
>>
>> root@sandbox:/usr/local/src/sipp.svn# ./sipp -sd
>> 2011-02-20 19:55:20:038 1298260520.038988: Missing argument for
>> param '-sd'.
>> Use 'sipp -h' for details.
>>
>
> You are asking sipp to dump a scenario but you are not telling which
> scenario must be dumped.
> It could be:
> sipp -sd uac
> or
> sipp -sd uas
> and so on.
>
>
>
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Message: 3
Date: Mon, 21 Feb 2011 15:52:36 -0000
From: "John McNamara" <john.mcnam...@emutex.com>
Subject: [Sipp-users] Timeout attribute on initial recv request
To: <sipp-users@lists.sourceforge.net>
Message-ID: <002901cbd1df$5f67a010$1e36e030$@mcnam...@emutex.com>
Content-Type: text/plain; charset="us-ascii"
Hi,
I can't get the timeout to work with a recv message in a simplified server
scenario such as the following:
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="SIPp Sequence">
<recv request="INVITE" timeout="1000" ontimeout="1" crlf="true" />
<label id="1" />
<nop>
<log message="Here 2"/>
<exec int_cmd="stop_now"/>
</nop>
</scenario>
When the scenario is run as follows it doesn't time out:
~/sipp.svn/sipp -sf timeout.xml --trace_logs
The scenario will time out if the -timeout option is passed from the
commandline
~/sipp.svn/sipp -sf timeout.xml --trace_logs -timeout 5
The SIPp version is 3.2 built from sipp.svn:
~/sipp.svn/sipp -v | grep SIPp
SIPp v3.2-TLS-PCAP, version unknown, built Feb 11 2011, 11:57:05.
Am I misunderstanding the intention of the timeout attribute in the above
example? Am I using it incorrectly?
John.
--
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Message: 4
Date: Tue, 22 Feb 2011 08:13:55 -0600
From: "Corcione, John" <john.corci...@hrblock.com>
Subject: [Sipp-users] Discarded message which can't be mapped & Can't
Hear Audio to called party.
To: "sipp-users@lists.sourceforge.net"
<sipp-users@lists.sourceforge.net>
Message-ID:
<dd2220c1bc39c04795acc71a9d93467b05307ec...@bcmsg01.hrbinc.hrblock.net>
Content-Type: text/plain; charset="us-ascii"
I am getting the following errors when trying to send a PCAP file stream to our
IVR Platform.
sipp: The following events occured:
2011-02-21 19:14:30:885 1298337270.885000: Aborted call with Call-ID
'1-4628@10.205.53.5'.
2011-02-21 19:14:30:886 1298337270.886000: Discarding message which can't
be mapped to a known SIPp call: <----
SIP/2.0 200 OK
From: <sip:sipp@10.205.53.5:5065>;tag=1
To:
<sip:78166650308@10.205.53.4:5060>;tag=624dd50c-13c4-4d630d8e-5b74ca2a-e95f244
Call-ID: 1-4628@10.205.53.5
CSeq: 2 BYE
Via: SIP/2.0/UDP 10.205.53.5:5065;branch=z9hG4bK-4628-1--1
Content-Length: 0
I think I May have the above corrected, However, would like an explanation of
the above if.
Also, I believe I am sending the correct RTP stream PCAP file. I am calling my
desk set - (Avaya IVR to Nortel IP Phone and AVAYA platform to Mobile phone
Call Flow). I see the RTP packets being sent but I do not hear the audio to
the called the called party. Please advise.
Please see Image attached.
-----Original Message-----
From: sipp-users-requ...@lists.sourceforge.net
[mailto:sipp-users-requ...@lists.sourceforge.net]
Sent: Friday, February 18, 2011 10:27 AM
To: sipp-users@lists.sourceforge.net
Subject: Sipp-users Digest, Vol 57, Issue 6
Send Sipp-users mailing list submissions to
sipp-users@lists.sourceforge.net
To subscribe or unsubscribe via the World Wide Web, visit
https://lists.sourceforge.net/lists/listinfo/sipp-users
or, via email, send a message with subject or body 'help' to
sipp-users-requ...@lists.sourceforge.net
You can reach the person managing the list at
sipp-users-ow...@lists.sourceforge.net
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Sipp-users digest..."
Today's Topics:
1. Re: Sipp-users Digest, Vol 57, Issue 5 (Corcione, John)
2. DTMF pcap files with payload type=127 (Shah, Nisha N. (Nisha))
3. Re: Example for register with authentication? (Gopalakrishnan A.N)
4. Re: Example for register with authentication? (mayamatakeshi)
----------------------------------------------------------------------
Message: 1
Date: Wed, 16 Feb 2011 10:02:26 -0600
From: "Corcione, John" <john.corci...@hrblock.com>
Subject: Re: [Sipp-users] Sipp-users Digest, Vol 57, Issue 5
To: "sipp-users@lists.sourceforge.net"
<sipp-users@lists.sourceforge.net>
Message-ID:
<dd2220c1bc39c04795acc71a9d93467b053065f...@bcmsg01.hrbinc.hrblock.net>
Content-Type: text/plain; charset="us-ascii"
We have to create SIP traffic toward our IVR platform. We need to send voice
energy to the platform once the call is established. Is there an example on
how to do this or has any one attempted this? I believe we need to send an
wave file once the call is established.
-----Original Message-----
From: sipp-users-requ...@lists.sourceforge.net
[mailto:sipp-users-requ...@lists.sourceforge.net]
Sent: Sunday, February 13, 2011 9:27 AM
To: sipp-users@lists.sourceforge.net
Subject: Sipp-users Digest, Vol 57, Issue 5
Send Sipp-users mailing list submissions to
sipp-users@lists.sourceforge.net
To subscribe or unsubscribe via the World Wide Web, visit
https://lists.sourceforge.net/lists/listinfo/sipp-users
or, via email, send a message with subject or body 'help' to
sipp-users-requ...@lists.sourceforge.net
You can reach the person managing the list at
sipp-users-ow...@lists.sourceforge.net
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Sipp-users digest..."
Today's Topics:
1. Re: while expecting '200' (index 5), received 'SIP/2.0 100
Trying (Sri Narayanan)
2. Re: while expecting '200' (index 5), received 'SIP/2.0 100
Trying (Sri Narayanan)
3. while expecting '200' (index 5), received 100 (Sri Narayanan)
4. expecting 200 received 100 (Sri Narayanan)
5. Re: expecting 200 received 100 (mayamatakeshi)
6. SIPP failed calls and RTP Flow (Anthony LELONG)
7. about scenarios of sipp (finandria audiary)
----------------------------------------------------------------------
Message: 1
Date: Wed, 9 Feb 2011 14:57:07 +0530
From: Sri Narayanan <sri.narayanan....@gmail.com>
Subject: Re: [Sipp-users] while expecting '200' (index 5), received
'SIP/2.0 100 Trying
To: mayamatakeshi <mayamatake...@gmail.com>
Cc: sipp-users-requ...@lists.sourceforge.net,
sipp-users@lists.sourceforge.net
Message-ID:
<aanlktinw3njyhmxmcthcuwepk14uh9yc1hkgkkzlt...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
<recv response="100" rrs="true" crlf="true" optional="true" />
<recv response="200" rtd="true">
This is what I have,I doubt the second stmt fail when it gets the 100
On Wed, Feb 9, 2011 at 2:48 PM, mayamatakeshi <mayamatake...@gmail.com> wrote:
>
> On Wed, Feb 9, 2011 at 6:12 PM, Sri Narayanan <sri.narayanan....@gmail.com>
> wrote:
>>
>> Aborting call on unexpected message for Call-Id
>> '9-14316@150.236.118.200': while expecting '200' (index 5), received
>> 'SIP/2.0 100 Trying
>>
>> How to make sipp ignore this message
>
> I would try this in the xml file:
> ? <recv response="100" optional="true"/>
>
> regards,
> takeshi
>
--
Regards,
SriNarayanan
------------------------------
Message: 2
Date: Wed, 9 Feb 2011 15:01:15 +0530
From: Sri Narayanan <sri.narayanan....@gmail.com>
Subject: Re: [Sipp-users] while expecting '200' (index 5), received
'SIP/2.0 100 Trying
To: mayamatakeshi <mayamatake...@gmail.com>
Cc: sipp-users-requ...@lists.sourceforge.net,
sipp-users@lists.sourceforge.net
Message-ID:
<AANLkTindU4cmPC1w93=uxed63qnej0x7qhgkek2bj...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
<recv response="100" rrs="true" crlf="true" optional="true" />
<recv response="200" rtd="true">
This is what I have,I doubt the second stmt fail when it gets the 100
Sri
------------------------------
Message: 3
Date: Wed, 9 Feb 2011 15:09:05 +0530
From: Sri Narayanan <sri.narayanan....@gmail.com>
Subject: [Sipp-users] while expecting '200' (index 5), received 100
To: sipp-users-requ...@lists.sourceforge.net,
sipp-users@lists.sourceforge.net
Message-ID:
<AANLkTi=agpfbnmt3msk+hsco+tzey0cehzknjdvg-...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
<recv response="100" rrs="true" crlf="true" optional="true" />
<recv response="200" rtd="true"> ------------>I guess the second line
is generating the error :while expecting '200' (index 5), received 100
--
Regards,
SriNarayanan
------------------------------
Message: 4
Date: Wed, 9 Feb 2011 15:23:56 +0530
From: Sri Narayanan <sri.narayanan....@gmail.com>
Subject: [Sipp-users] expecting 200 received 100
To: viswavardhanreddy karna <viswavardhanre...@gmail.com>,
sipp-users-requ...@lists.sourceforge.net,
sipp-users@lists.sourceforge.net
Message-ID:
<AANLkTi=enrix_2xmpwwsomk62grj_srkgx9kzwbiy...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi Karna,
Actually there is an action tag follows :-)
<recv response="100" rrs="true" crlf="true" optional="true" />
<recv response="200" rtd="true"/>
<action optional="true">
<ereg regexp="fid=([[:alnum:]]*)_([[:alnum:]]*)"
search_in="hdr" header="Contact:" check_it="true" assign_to="1" />
<ereg
regexp="bekey=([[:alnum:]]*)-([[:alnum:]]*)%([[:alnum:]]*).([[:alnum:]]*).([[:alnum:]]*).([[:alnum:]]*)"
search_in="hdr" header="Contact:" check_it="true" assign_to="2" />
</action>
</recv>
--
Regards,
SriNarayanan
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------------------------------
Message: 5
Date: Wed, 9 Feb 2011 19:00:43 +0900
From: mayamatakeshi <mayamatake...@gmail.com>
Subject: Re: [Sipp-users] expecting 200 received 100
To: Sri Narayanan <sri.narayanan....@gmail.com>
Cc: Sipp <sipp-users@lists.sourceforge.net>
Message-ID:
<AANLkTi=Uv2FUgViPHZsLUc6srEW=fGBvf6qebY=o_...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
On Wed, Feb 9, 2011 at 6:53 PM, Sri Narayanan
<sri.narayanan....@gmail.com>wrote:
> Hi Karna,
> Actually there is an action tag follows :-)
>
> <recv response="100" rrs="true" crlf="true" optional="true" />
>
>
>
> <recv response="200" rtd="true"/>
>
> <action optional="true">
> <ereg regexp="fid=([[:alnum:]]*)_([[:alnum:]]*)"
> search_in="hdr" header="Contact:" check_it="true" assign_to="1" />
> <ereg
> regexp="bekey=([[:alnum:]]*)-([[:alnum:]]*)%([[:alnum:]]*).([[:alnum:]]*).([[:alnum:]]*).([[:alnum:]]*)"
> search_in="hdr" header="Contact:" check_it="true" assign_to="2" />
> </action>
>
> </recv>
The xml is malformed. You are closing the receive node at this point:
<recv response="200" rtd="true"/>
and
closing again at
</recv>
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Message: 6
Date: Thu, 10 Feb 2011 11:12:35 +0100
From: Anthony LELONG <lelo...@free.fr>
Subject: [Sipp-users] SIPP failed calls and RTP Flow
To: sipp-users@lists.sourceforge.net
Message-ID: <1297332755.4d53ba1317...@imp.free.fr>
Content-Type: text/plain; charset=ISO-8859-1
Hi everybody.
I've seen in the sipp documentation that we can get statistics about failed
calls. I know that counter concerns SIP errors. However, I wonder if that
counter deals with RTP errors too?
I mean, if a RTP packet is lost, is it considered as a failed call? Does it
increment a counter and, if so, which one?
Thanks a lot!
Anthony.
------------------------------
Message: 7
Date: Sun, 13 Feb 2011 22:26:38 +0700
From: finandria audiary <audi...@gmail.com>
Subject: [Sipp-users] about scenarios of sipp
To: sipp-users@lists.sourceforge.net
Message-ID:
<aanlktinp7o2rxyqka_9hko0afanjtyegopg55pwaz...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
can u help me, to send the scenarios of sipp?
specially for IMS..
thank's a lot.. :)
------------------------------
------------------------------------------------------------------------------
The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE:
Pinpoint memory and threading errors before they happen.
Find and fix more than 250 security defects in the development cycle.
Locate bottlenecks in serial and parallel code that limit performance.
http://p.sf.net/sfu/intel-dev2devfeb
------------------------------
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
End of Sipp-users Digest, Vol 57, Issue 5
*****************************************
------------------------------
Message: 2
Date: Thu, 17 Feb 2011 22:23:27 -0700
From: "Shah, Nisha N. (Nisha)" <nns...@avaya.com>
Subject: [Sipp-users] DTMF pcap files with payload type=127
To: <sipp-users@lists.sourceforge.net>
Message-ID:
<e518af19dcd30e4eab872ed6a08c4af902475...@306181anex2.global.avaya.com>
Content-Type: text/plain; charset="us-ascii"
Hello,
Does anyone know how I can generate DTMF digit pcap files with payload
type =127? The digit pcap files that I downloaded from the SIPp
website have a payload type=101, and currently I need to test with the
127 payload type.
Any help is appreciated.
Thanks,
Nisha
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Message: 3
Date: Fri, 18 Feb 2011 21:36:15 +0530
From: "Gopalakrishnan A.N" <sai...@gmail.com>
Subject: Re: [Sipp-users] Example for register with authentication?
To: mayamatakeshi <mayamatake...@gmail.com>
Cc: sipp-users@lists.sourceforge.net
Message-ID:
<aanlktin86vmnwc6wuqflb2_hxoe_vap8mtzl0eh8k...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
Thanks for the script. I am the basic learner of the SIPp. I am using SIPp
with Trixbox, as per your script i tried to execute and results to a error
like this,
2011-02-18 21:33:19:375 1298044999.375658: Unknown search_in value
var.
Would appreciate for your feedback.
On Mon, Feb 7, 2011 at 6:38 AM, mayamatakeshi <mayamatake...@gmail.com>wrote:
>
> On Mon, Feb 7, 2011 at 2:06 AM, Thomas Elsgaard <thomas.elsga...@gmail.com
> > wrote:
>
>> Hello everybody
>>
>> I am trying to get started with SIPP, but i am a little bit stucked in
>> building an scenario file for registering a user with authentication..
>>
>> Do anybody have an example scenario file i can try with ? maybe including
>> reading the username and password from a csv file
>>
>>
>
> Use the attached file.
> You might find other scenarios searching the mailing list archives.
>
> regards,
> takeshi
>
>
> ------------------------------------------------------------------------------
> The modern datacenter depends on network connectivity to access resources
> and provide services. The best practices for maximizing a physical server's
> connectivity to a physical network are well understood - see how these
> rules translate into the virtual world?
> http://p.sf.net/sfu/oracle-sfdevnlfb
> _______________________________________________
> Sipp-users mailing list
> Sipp-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>
>
--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
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Message: 4
Date: Sat, 19 Feb 2011 01:26:47 +0900
From: mayamatakeshi <mayamatake...@gmail.com>
Subject: Re: [Sipp-users] Example for register with authentication?
To: "Gopalakrishnan A.N" <sai...@gmail.com>
Cc: sipp-users@lists.sourceforge.net
Message-ID:
<AANLkTim1Dh2hPXv8FFbsLRvA2rPLg=+o-p+el1vm2...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello,
this means you are using an old version of SIPp.
Get the latest and build it from source.
On Sat, Feb 19, 2011 at 1:06 AM, Gopalakrishnan A.N <sai...@gmail.com>wrote:
> Hi,
>
> Thanks for the script. I am the basic learner of the SIPp. I am using SIPp
> with Trixbox, as per your script i tried to execute and results to a error
> like this,
>
> 2011-02-18 21:33:19:375 1298044999.375658: Unknown search_in value
> var.
>
> Would appreciate for your feedback.
>
> On Mon, Feb 7, 2011 at 6:38 AM, mayamatakeshi <mayamatake...@gmail.com>wrote:
>
>>
>> On Mon, Feb 7, 2011 at 2:06 AM, Thomas Elsgaard <
>> thomas.elsga...@gmail.com> wrote:
>>
>>> Hello everybody
>>>
>>> I am trying to get started with SIPP, but i am a little bit stucked in
>>> building an scenario file for registering a user with authentication..
>>>
>>> Do anybody have an example scenario file i can try with ? maybe including
>>> reading the username and password from a csv file
>>>
>>>
>>
>> Use the attached file.
>> You might find other scenarios searching the mailing list archives.
>>
>> regards,
>> takeshi
>>
>>
>> ------------------------------------------------------------------------------
>> The modern datacenter depends on network connectivity to access resources
>> and provide services. The best practices for maximizing a physical
>> server's
>> connectivity to a physical network are well understood - see how these
>> rules translate into the virtual world?
>> http://p.sf.net/sfu/oracle-sfdevnlfb
>> _______________________________________________
>> Sipp-users mailing list
>> Sipp-users@lists.sourceforge.net
>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>
>>
>
>
> --
> Thank you with regards,
> Gopalakrishnan A.N.
> VoIP call - sip:sai...@gtalk2voip.com
>
>
>
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The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE:
Pinpoint memory and threading errors before they happen.
Find and fix more than 250 security defects in the development cycle.
Locate bottlenecks in serial and parallel code that limit performance.
http://p.sf.net/sfu/intel-dev2devfeb
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Index, Search & Analyze Logs and other IT data in Real-Time with Splunk
Collect, index and harness all the fast moving IT data generated by your
applications, servers and devices whether physical, virtual or in the cloud.
Deliver compliance at lower cost and gain new business insights.
Free Software Download: http://p.sf.net/sfu/splunk-dev2dev
------------------------------
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Index, Search & Analyze Logs and other IT data in Real-Time with Splunk
Collect, index and harness all the fast moving IT data generated by your
applications, servers and devices whether physical, virtual or in the cloud.
Deliver compliance at lower cost and gain new business insights.
Free Software Download: http://p.sf.net/sfu/splunk-dev2dev
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