Dear all
After act as my leader's advice.
I try to change to using Window XP and everything work finely.
I think my installation process in Ubuntu (10.10) has problem ( may be
library.....)
So if you encounter the same problem. Please try as i do.
And a final problem ( only occur when you using the long xml scenario ) is
synchronization between READ cmd in Asterisk server and Pause <time> after
pressing a button....
Thank so much.
On Tue, Oct 19, 2010 at 4:00 PM, Mr Trung ND <nguyentrun...@gmail.com>wrote:
> Dear all.
> After comparing between the same digit sent in one session to the SIP
> server (asterisk) by X-lite (Softfone) and SIPp i found some differences
> here:
> the time duration of continuous packet in one time press (in my case, that
> is number 7 )
> X-lite : 160 - 320 - 480 - 640 -800.....(adding 160 after one packet)
> SIPp : 0 - 320 - 640 - 1280..... ( x2 after one packet)
>
> * *I think that's the reason why Asterisk doesn't recognize the pressed
> button.
> Any ideal on fix the duration ( as refer at 3.5 at
> this<http://www.ietf.org/rfc/rfc2833.txt> it's timestamps
> of packets)
>
> many thank, if you know , please help...
>
>
>
>
> On Tue, Oct 19, 2010 at 8:59 AM, Mr Trung ND <nguyentrun...@gmail.com>wrote:
>
>> I do step by step following this post
>>
>>
>> http://ismellpackets.com/2009/06/02/answer-to-freds-secret-packet-challenge-part-1/
>>
>>
>> <http://ismellpackets.com/2009/06/02/answer-to-freds-secret-packet-challenge-part-1/>by
>> using Wireshark and X-lite (which already connected well to Asterisk
>> server).
>> I press any button while calling to Asterisk server and using the
>> Wireshark to capture all the packet, analyse the packet has brief info like
>> "
>>
>> 1000 12.522205 192.168.0.177 192.168.0.178 RTP EVENT Payload type=RTP
>>> Event, DTMF Seven 7
>>
>>
>> "
>> it's about 13-14 packet in totall .
>>
>> After convert to a pcap file. It still not work. Anybody who's ever done
>> like this, Please help and guide me.
>>
>> Thank All.
>>
>>
>> On Mon, Oct 18, 2010 at 11:14 AM, Mr Trung ND <nguyentrun...@gmail.com>wrote:
>>
>>> I've manage to using the recorded from packet via a softphone (like
>>> X-lite) to a/some pcap file to replay it instead of default files.
>>> Is any ideals or experiences on this issue, Please help and share.
>>> Thank you so much.
>>> Brs
>>>
>>>
>>> On Thu, Oct 14, 2010 at 6:53 PM, Mr Trung ND <nguyentrun...@gmail.com>wrote:
>>>
>>>> Dear all SIPp Users.
>>>>
>>>> I'm a newbie, and using the SIPp since about week ago.
>>>> On the purpose to stress the Asterisk Server i use a different PC that
>>>> run SIPp with PCAP play ability in the same LAN.
>>>> In our scenarios i must send consequential DTMF digits to SIP server
>>>> like this '7' digit
>>>>
>>>>> <!-- Play an out of band DTMF '7'
>>>>> -->
>>>>
>>>> <nop>
>>>>
>>>> <action>
>>>>
>>>> <exec
>>>>> play_pcap_audio="/home/trungnd/sipp.svn/pcap/dtmf_2833_7.pcap"/>
>>>>
>>>> </action>
>>>>
>>>> </nop>
>>>>
>>>>
>>>> I've installed Wireshark on two side PC to capture RTP packets.
>>>> I also using a softphone (x-lite) (on client side) to branch into the
>>>> same scenario to compare.
>>>> At the both case the Wireshark recognized there are DTMF digits send
>>>> out SIPp's PC and come in SIPp server (two side)
>>>>
>>>> 1102 15.812024 192.168.0.175 192.168.0.178 RTP EVENT
>>>>>> Payload type=RTP Event, DTMF Seven 7
>>>>>
>>>>>
>>>>>>
>>>> And after reference an user from mail i've already edit the "rtp.conf"
>>>> in SIP server ( lower the start port, and higher end port) to ensure the
>>>> incoming port are in range.
>>>> *But when test with a simple scenario, the softphone call to sip server
>>>> and the server recognized which digit is pressed and branch, while the
>>>> sipp
>>>> call to sip server, the server do not know in most case (but not all) .
>>>> *
>>>> I'm in urgent case because of death line is coming. If you have any
>>>> experiences .*Please advice.*
>>>>
>>>>
>>>> **Thank you so much.
>>>>
>>>
>>>
>>
>
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