Hello,
First I'd like to say thank you for sipp. I have been having a rough time
trying to send dtmf with sipp. At first I was using sippy_cup but decided I'd
try this in sipp. I was having similar issues with sippy_cup. I have been
trying to replay pcap files and the included dtmf tones in the /pcap directory
for sipp and some captures I've got with tcpdump. I built sipp from source with
pcap support SIPp v3.5.1-PCAP-RTPSTREAM. I'm running FreePBX 14.0.3.6 from
raspbx on a rasberry pi for testing. I'm running sipp on the same host as
FreePBX also.
Goal: Test ivr with 5-6 dtmf tones for load and errors.
In the cdr reports I always see sipp calling from the destination "s
[from-trunk]" in my cdr reports. I know that by default sipp dials s, can I
change that? I can see the dtmf tones in the full log and asterisk cli like
below. I have also tried all the different dtfm modes in the Settings>Advanced
Settings and the trunk details inside freepbx.
[2018-06-22 13:06:32] DTMF[9942][C-00000028]: channel.c:4040 __ast_read: DTMF
end '1' received on SIP/127.0.1.1-00000028, duration 0 ms
[2018-06-22 13:06:32] DTMF[9942][C-00000028]: channel.c:4099 __ast_read: DTMF
end accepted without begin '1' on
SIP/127.0.1.1-00000028
[2018-06-22 13:06:32] DTMF[9942][C-00000028]: channel.c:4110 __ast_read: DTMF
end passthrough '1' on SIP/127.0.1.1-00000028
[2018-06-22 13:06:35] DTMF[9959][C-00000029]: channel.c:4040 __ast_read: DTMF
end '1' received on SIP/127.0.1.1-00000029, duration 0 ms
[2018-06-22 13:06:35] DTMF[9959][C-00000029]: channel.c:4099 __ast_read: DTMF
end accepted without begin '1' on
SIP/127.0.1.1-00000029
Scenario below
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
[11/72]
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- 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="1000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
I've tried this in a scenario also. Thinking I needed a pause between the tones.
</send>
<pause milliseconds="15000" />
<nop>
<action>
<exec play_pcap_audio="pcap\dtmf_2833_1.pcap" />
</action>
</nop>
<pause milliseconds="750" />
<nop>
<action>
<exec play_pcap_audio="pcap\dtmf_2833_0.pcap" />
</action>
</nop>
<pause milliseconds="750" />
<nop>
<action>
<exec play_pcap_audio="pcap\dtmf_2833_0.pcap" />
</action>
</nop>
<pause milliseconds="750" />
<nop>
<action>
<exec play_pcap_audio="pcap\dtmf_2833_5.pcap" />
</action>
</nop>
<pause milliseconds="750" />
<nop>
<action>
<exec play_pcap_audio="pcap\dtmf_2833_0.pcap" />
</action>
</nop>
<pause milliseconds="40000" />
<send>
I have been playing some pcaps that I got via tcpdump and the included dtmf
tones in the pcap directory. I can see the dtfm tones in the call flow in
wireshark and hear them. I also separated both legs of the call (because both
legs did not work), to try just the part from the trunk with the tones. I got
the same results as having both legs of the call in the pcap. I even tried some
pcaps from wiresharks site and got some results. I can see asterisk responding
with SayAlpha in the cdr reports and the logs.
To acheive this do I need to patch sipp with the inband dtfm patch here or some
other patch? https://sourceforge.net/p/sipp/patches/50/ I've tried and I can't
compile it after the patch.
How can I get sipp calling my ivr and getting the dtmf tones accepted? Am I
using the wrong tool for this? Is there anything better? I was thinking about
making a script to just make the calls like normal. Like this maybe?
https://obrienlabs.net/automate-asterisk-to-auto-dial-a-number-for-testing/ If
this has been asked on the list before I'm sorry in advance, I tried searching
it. Thanks for taking the time to read this.
Have a good day.
Reese
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