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

[Sakharam Thorat]

This may be usefull. http://sourceforge.net/p/sipp/mailman/message/22371226/

Best Regards,Sakharam Thorat.From: manishanand.t at 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.thorat at outlook.com
CC: saurabh.shah at matrixcomsec.com; discussion at 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.thorat at 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: manishanand.t at gmail.com
Date: Mon, 9 Mar 2015 12:56:51 +0530
To: saurabh.shah at matrixcomsec.com
CC: discussion at 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:+4032011234 at im.com>;tag=1P-Early-Media: sendonlyAllow: INVITE,ACK,PRACK,BYECall-ID: 1-14036 at 127.0.0.1From: <sip:+4032014321 at 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.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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Thanks and Regards

Manish Anand


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Thanks and Regards

Manish Anand
 		 	   		  
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