HI:
First Thank catalina very much!
Now, to exist some new questions, i want to achieve different calling
number <app:ds:correspond> correspond to different DTMF
for exemple:
csv
SEQUENTIAL
123;8888
123;9999
If the calling number is 8888, go to xml
<!-- Play an out of band DTMF '1' -->
<action>
<exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
</action>
If the calling number is 9999, go to xml
<!-- Play an out of band DTMF '2' -->
<action>
<exec play_pcap_audio="pcap/dtmf_2833_2.pcap"/>
</action>
Is action set at variable?
Best Regards | parallel winding <app:ds:parallel%20winding>
Christy Gu
<blocked::http://www.ti-net.com.cn/>
_____
发件人: 安静波 [mailto:[email protected]]
发送时间: 2009年4月17日 14:34
收件人: catalina oancea
抄送: Sipp
主题: RE: [Sipp-users] [Need Help] sipp pcapplay feature does not work.
Hi,
The issue has benn resoved. The option -mp of sipp is uesd to set the
local RTP echo port number. Default is 6000. Asterisk's rtp.conf is
configured as : rtpstart=10000 rtpend=20000, after change rtpstart=5000, The
pcapplay feature works. BTW, it seems that the local RTP echo port number
set by -mp option must less than 10000.
Any way, Thank catalina very much!
Best Regards,
Aaron An
_____
From: [email protected]
To: [email protected]
Date: Fri, 17 Apr 2009 03:29:10 +0000
CC: [email protected]
Subject: Re: [Sipp-users] [Need Help] sipp pcapplay feature does not work.
Hi,
Thanks a lot for the idea, I set rtp debug on in asterisk and found that
no rtps received from the sipp client. I also cap udp packets by tcpdump on
sipp client, no rtp packets.
By the way, iptables was stopp. So It must be the issue of sipp client.
The error file generated by sipp shows that there is no error. Is any
idea to see more debug info?
The senario screen is as belew:
------------------------------ Scenario Screen -------- [1-9]: Change Screen
--
Call-rate(length) Port Total-time Total-calls Remote-host
1.0(0 ms)/1.000s 5061 19.00 s 1 10.10.10.57:5060(UDP)
Call limit reached (-m 1), 1.000 s period 1 ms scheduler resolution
1 calls (limit 300) Peak was 1 calls, after 1 s
0 Running, 1 Paused, 0 Woken up
0 dead call msg (discarded) 0 out-of-call msg (discarded)
3 open sockets
412 Total RTP pckts sent 8.316 last period RTP rate (kB/s)
Many Thansk!
Best Regards,
Aaron An
> Date: Fri, 17 Apr 2009 01:50:19 +0300
> Subject: Re: [Sipp-users] [Need Help] sipp pcapplay feature does not work.
> From: [email protected]
> To: [email protected]
> CC: [email protected]
>
> I used the g711a.pcap file to send rtp on centos and it worked for me.
> If you capture outgoing packages(tcpdump or similar), can you see any
> outgoing rtp from the host running sipp? (if you are using asterisk,
> you can also use rtp debug in asterisk CLI to see if any rtp is sent)
> You can also look at the error file generated by sipp(you can find it
> in the current directory after running sipp).
>
>
>
> 2009/4/16 <[email protected]>:
> > Hi all,
> >
> > I am a newer in using "sip" to test soft-pbx Asterisk, when I install
> > "sip" at Windows XP platform it works well and i can hear the tone
played by
> > the pcapplay feature. But there is an issue when sipp is install at
Linux,
> > nothing played as I use the same xml file the same csv file and the
> > same command option. Neither g711a.pcap nor DTMF '1' can be heared.
> >
> >
> >
> > Here is my environment:
> >
> > 1. CentOS linux
> >
> > 2. libnet, libpcap, libpcap-devel installed
> >
> > 3. "make pcapplay" to build sip
> >
> > 4. Part of xml file: The same xml file workes well in windows.
> >
> > <!-- Play a pre-recorded PCAP file (RTP stream) -->
> >
> > <nop>
> >
> > <action>
> >
> > <exec play_pcap_audio="pcap/g711a.pcap"/>
> >
> > </action>
> >
> > </nop>
> >
> >
> >
> > <!-- Pause 8 seconds, which is approximately the duration of the -->
> >
> > <!-- PCAP file -->
> >
> > <pause milliseconds="8000"/>
> >
> >
> >
> > <!-- Play an out of band DTMF '1' -->
> >
> >
> >
> > <nop>
> >
> > <action>
> >
> > <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
> >
> > </action>
> >
> > </nop>
> >
> >
> >
> > <pause milliseconds="3000"/>
> >
> > 5. command:
> >
> > sipp -sf sn.xml -inf cn.csv -p 5061 -i 10.10.10.144 -m 1 -l 300 -r 1
> > 10.10.10.57:5060 -trace_screen -trace_err
> >
> >
> >
> > Any ideas? Many thanks.
> >
> >
> >
> >
> >
> > Best Regards,
> >
> > Aaron An
> >
> >
> >
> >
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> >
> >
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------------------------------------------------------------------------------
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around Java (TM) technology - register by April 22, and save
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300 plus technical and hands-on sessions. Register today.
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