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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