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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