Hello, I was able to solve this by creating a misc application and pointing that at the ivr and just dialing it. Hope this helps someone else.
Thanks! On Mon, Jun 25, 2018 at 09:10:26AM -0500, Reese wrote: > 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 > > ------------------------------------------------------------------------------ > Check out the vibrant tech community on one of the world's most > engaging tech sites, Slashdot.org! http://sdm.link/slashdot > _______________________________________________ > Sipp-users mailing list > Sipp-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/sipp-users ------------------------------------------------------------------------------ Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
