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 it's timestamps of packets)
many thank, if you know , please help...
On Tue, Oct 19, 2010 at 8:59 AM, Mr Trung ND <firstname.lastname@example.org>
I do step by step following this post
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.
On Mon, Oct 18, 2010 at 11:14 AM, Mr Trung ND <email@example.com>
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.
On Thu, Oct 14, 2010 at 6:53 PM, Mr Trung ND <firstname.lastname@example.org>
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' -->
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.