This may be usefull. http://sourceforge.net/p/sipp/mailman/message/22371226/
Best Regards,Sakharam Thorat.From: manishanan...@gmail.com
Date: Tue, 10 Mar 2015 18:49:15 +0530
Subject: Re: [SIPForum-discussion] Load call test for sip audio calls with DTMF
To: sakharam.tho...@outlook.com
CC: saurabh.s...@matrixcomsec.com; discuss...@sipforum.org
Hi Sakharam,
This link contain A party script, I want B party script. If you have please
share with me .
Thanks,
Manish Anand
On Tue, Mar 10, 2015 at 6:40 PM, Sakharam Thorat <sakharam.tho...@outlook.com>
wrote:
Take look at http://www.mobicents.org/mss/ssf/sf-flow-api/uac.xml it were
worked for me .
Best Regards,Sakharam Thorat.
From: manishanan...@gmail.com
Date: Mon, 9 Mar 2015 12:56:51 +0530
To: saurabh.s...@matrixcomsec.com
CC: discuss...@sipforum.org
Subject: Re: [SIPForum-discussion] Load call test for sip audio calls with
DTMF
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/2.0Via: SIP/2.0/UDP
127.0.0.1:25060;branch=z9hG4bK-14036-1-3CSeq: 2 PRACKTo:
<sip:+4032011...@im.com>;tag=1P-Early-Media: sendonlyAllow:
INVITE,ACK,PRACK,BYECall-ID: 1-14036@127.0.0.1From:
<sip:+4032014...@mt.com>;tag=1Max-Forwards: 70RAck: 1 INVITERoute:
<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.s...@matrixcomsec.com>
wrote:
Use SIPP.
On Fri, Feb 20, 2015 at 3:38 PM, Antonio Manchon Romero
<antoniomanchonrom...@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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Thanks and Regards
Manish Anand
_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
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Thanks and Regards
Manish Anand
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