Hi,
do you have RTP ports number uder 9999 ? It is some limitation of SIPp  
- if you have port higher than 9999 the RTP stream doesn´t play.
And next advice...the DTMF pcap files have some timestamps and if you  
play for example DTMF number 7 (higher timestamp) and the 6 (lower  
timestamp) it won´t work...you have to play 1,2...4...6...and so  
on...Or you can record your own consecution of DTMF numbers to pcap  
file.
I hope I am right in my advices...this advices I learn from reading  
SIPp mailing list "forum" ... Somebody solve there some problems with  
DTMF...try google and I am sure you will find it... (try read  
http://www.mail-archive.com/sipp-users@lists.sourceforge.net/msg03906.html)
PS: Excuse my english...I am Czech.

Ondrej Cakan


Cituji Mr Trung ND <nguyentrun...@gmail.com>:

> 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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