[SIPForum-discussion] Load call test for sip audio calls with DTMF

[Manish Anand]

Hi All,


I am facing one issue, when i am sending prack message through sipp tool,
In Sipp message log message is printing correctly like

PRACK sip:127.0.0.1:35060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:25060;branch=z9hG4bK-14036-1-3
CSeq: 2 PRACK
To: <sip:+4032011234 at im.com>;tag=1
P-Early-Media: sendonly
Allow: INVITE,ACK,PRACK,BYE
Call-ID: 1-14036 at 127.0.0.1
From: <sip:+4032014321 at mt.com>;tag=1
Max-Forwards: 70
RAck: 1 INVITE
Route: <sip:127.0.0.1:5060;lr>
Contact: <sip:127.0.0.1:25060>
Content-Length:     0

but in sip servlet in am getting error in doPrack method.

*java.lang.IllegalArgumentException: cseq number must be positive*


Please suggest where i am doing mistake.



On Sat, Mar 7, 2015 at 1:01 PM, Saurabh Shah <saurabh.shah at matrixcomsec.com>
wrote:

> Use SIPP.
>
> On Fri, Feb 20, 2015 at 3:38 PM, Antonio Manchon Romero <
> antoniomanchonromero at yahoo.es> wrote:
>
>> Hi everybody,
>>
>> I´m looking for a free tool to test a new environment. It would be for
>> audio sip calls only. Codecs: G711, G729 and dtmf capabilities would be
>> required.
>>
>> This software would be able to launch and maintain automatically several
>> calls. With this software I would like to launch calls progressively to the
>> new environment to test system delays and performance.
>>
>> After setup, software would have to send two DMTF sequences in
>> communication with an IVR. So it would have to send first sequence of
>> digits and after a known delay the second one.
>>
>> Thanks for your help,
>>
>> Antonio
>>
>>
>>
>> _______________________________________________
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>>
>
>
> --
> Regards,
> Saurabh Shah
>
> _______________________________________________
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> TO UNSUBSCRIBE, or edit your delivery options, please visit
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>
>


-- 
Thanks and Regards

Manish Anand
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