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SOLVED Sipp dtmf tones not recognised


(Bob) #1

Hello,

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 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.

[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" ?> <![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

]]>
                                                                                                     [11/72]
<![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

]]>
<![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

]]>

I’ve tried this in a scenario also. Thinking I needed a pause between the tones.

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 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 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 or some other patch? 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? Thanks for taking the time to read this.

Have a good day.


(Bob) #2

I was able to solve this by creating a misc application and point that at my ivr. Maybe this will help someone else.

Peace


(system) #3

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