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

[Pradyumna Padhee]

Hi Manish,

Please ensure that while,the request reaching the server,It's
MAX_FORWARD,is not zero(0).
If MAX-FORWARD becomes zero,it will send 483 Too Many Hops.

Thanks,
Pradyumna

On Tue, Mar 10, 2015 at 10:11 AM, Manish Anand <manishanand.t at gmail.com>
wrote:

> Hi Ranjit/Pradyumna,
>
> Thanks for you solving input, Now working fine this scenario. Now i am
> getting one more error like
>
>
> 2015-03-09    18:34:02:644    1425906242.644610: Aborting call on
> unexpected message for Call-Id '1-24978 at 169.144.33.14': while expecting
> '200' (index 4), received 'SIP/2.0 483 Too Many Hops.
>
> Any idea on this.
>
> Thanks,
>
> Manish Anand
>
>
>
>
>
>
> On Tue, Mar 10, 2015 at 12:37 AM, Pradyumna Padhee <pkumarp1987 at gmail.com>
> wrote:
>
>> Hi,
>>
>> PRACK should contain the RAck as (Integer part of Rseq came in 18x
>> response and Cseq of INVITE)[image: Inline image 1]
>>
>> Please try this.
>>
>> Thanks,
>> Pradyumna
>>
>>
>>
>>
>>
>>
>> On Mon, Mar 9, 2015 at 12:56 PM, Manish Anand <manishanand.t at gmail.com>
>> wrote:
>>
>>> 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
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>>>>> 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&Regards
>> Pradyumna Padhee
>> mob:-9711176064
>>
>
>
>
> --
> Thanks and Regards
>
> Manish Anand
>



-- 
Thanks&Regards
Pradyumna Padhee
mob:-9711176064
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