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

Mon Mar 16 09:26:28 UTC 2015

Hi,

I am facing one issue, when I am running the sip script on test server,
prack 200 ok response is getting timed out, but I can see Invite 200 Ok
response which I am sending after prack 200 ok response is getting
successful.

Same scenario is working in dev environment.

Please help me..

Thanks,

Manish Anand

On Fri, Mar 13, 2015 at 5:53 PM, Manish Anand <manishanand.t at gmail.com>
wrote:

> Hi,
>
> I am getting one issue, prack 200 ok response it not is going to party A.
> getting time out. any input why can be the issue.
>
> Thanks,
>
> Manish Anand
>
>
> On Thu, Mar 12, 2015 at 4:18 PM, Manish Anand <manishanand.t at gmail.com>
> wrote:
>
>> Hi,
>>
>> After pack and ack, when A party is sending bye request to B Party, in
>> sip servlet I am getting below error. Any idea on this..
>>
>> java.lang.AssertionError
>>  at
>> com.bea.wcp.sip.engine.server.SipServletRequestImpl.createResponse(SipServletRequestImpl.java:1282)
>>  at
>> com.bea.wcp.sip.engine.SipServletRequestAdapter.createResponse(SipServletRequestAdapter.java:295)
>>  at javax.servlet.sip.SipServlet.notHandled(Unknown Source)
>>  at javax.servlet.sip.SipServlet.doBye(Unknown Source)
>>
>> Thanks,
>>
>> Manish Anand
>>
>> On Thu, Mar 12, 2015 at 11:55 AM, Manish Anand <manishanand.t at gmail.com>
>> wrote:
>>
>>> Dear friend,
>>>
>>> After pack and ack, when A party is sending bye request to B Party, in
>>> sip servlet I am getting below error. Any idea on this..
>>>
>>> java.lang.AssertionError
>>>  at
>>> com.bea.wcp.sip.engine.server.SipServletRequestImpl.createResponse(SipServletRequestImpl.java:1282)
>>>  at
>>> com.bea.wcp.sip.engine.SipServletRequestAdapter.createResponse(SipServletRequestAdapter.java:295)
>>>  at javax.servlet.sip.SipServlet.notHandled(Unknown Source)
>>>  at javax.servlet.sip.SipServlet.doBye(Unknown Source)
>>>
>>>
>>>
>>> On Tue, Mar 10, 2015 at 7:00 PM, Sakharam Thorat <
>>> sakharam.thorat at outlook.com> wrote:
>>>
>>>>
>>>> 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/2.0
>>>> 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
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> This is the SIP Forum discussion mailing list
>>>> TO UNSUBSCRIBE, or edit your delivery options, please visit
>>>> http://sipforum.org/mailman/listinfo/discussion
>>>> Post to the list at discussion at sipforum.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Saurabh Shah
>>>>
>>>> _______________________________________________
>>>> This is the SIP Forum discussion mailing list
>>>> TO UNSUBSCRIBE, or edit your delivery options, please visit
>>>> http://sipforum.org/mailman/listinfo/discussion
>>>> Post to the list at discussion at sipforum.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Thanks and Regards
>>>>
>>>> Manish Anand
>>>>
>>>> _______________________________________________ This is the SIP Forum
>>>> discussion mailing list TO UNSUBSCRIBE, or edit your delivery options,
>>>> please visit http://sipforum.org/mailman/listinfo/discussion Post to
>>>> the list at discussion at sipforum.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Thanks and Regards
>>>>
>>>> Manish Anand
>>>>
>>>
>>>
>>>
>>> --
>>> Thanks and Regards
>>>
>>> Manish Anand
>>>
>>
>>
>>
>> --
>> Thanks and Regards
>>
>> Manish Anand
>>
>
>
>
> --
> Thanks and Regards
>
> Manish Anand
>

-- 
Thanks and Regards

Manish Anand
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20150316/c262e3d6/attachment-0002.html>