Re: [Sipp-users] [Urgent] Transfer DTMF digits using PCAP to Asterisk Server.

[Mr T. ND <ngu...@gm...>]

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 <ngu...@gm...>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 <ngu...@gm...>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 <ngu...@gm...>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.
>>>
>>
>>
>