Hi All,

Thought I would share my solution to this DTMF problem: it turns out my ISP was capping my bandwidth & dropping packets to keep the connection & 1Mbps, so the experienced DTMF loss was actually packets being discarded.

On my way to this discovery I tested Freeswitch & DTMF quite thoroughly & never actually found any problems even at hundreds of concurrent calls. Here is how I tested, who knows this might be useful to someone:

I used SIPp to generate calls & a Python script to log the received DTMF digits
SIPp command line:

sipp -sf dtmfSenario.xml -d 10000 -s 451 -l 96 -mp 5606 -i xxx.xxx.xxx.xxx

dtmfSenario.xml below
Dialplan:

<extension name="test_dtmf_capture_test">
      
      <condition field="destination_number" expression="(^100100$)">
        <action application="answer"/>
        <action application="python" data="writeDtmfStats"/>
      </condition>
    </extension>

Python:

import sys
from freeswitch import *

def get_number(session,invalid,num=20):
        digits = session.getDigits(num, "", 15000)

        consoleLog("info","Got '%s' digits from user.\n" % digits)
        if digits == '':
                # Invalid call
                if invalid == 3:
                        consoleLog("info","Three invalid attempts!!\n")
                        session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav")
                        session.hangup()
                        sys.exit(0)
                else:
                        session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav")
                        get_number(session,invalid + 1)
        else:
                consoleLog("info","Got a valid number: %s, proceeding...\n" % digits)
                return digits

def handler(session, args):
    session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav")
    numberToDial = get_number(session,2,num=10)
    consoleLog('info','Got 10 DTMF digits. Writing "1" to file...\n')
    fo = open('/tmp/dtmfData.csv','a')
    fo.write('"1"\n')
    fo.close()
    # Do some stuff & wait for SIPP to hangup
    session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav")
    session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav")
    return

DTMF senario file:

# cat dtmfSenario.xml
<?xml version="1.0" encoding="ISO-8859-1"?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

















<scenario name="UAC with media">


<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=[call_number]
      To: sut <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 18 100
      a=rtpmap:18 G729/8000
      a=fmtp:18 annexb=no
      a=rtpmap:100 telephone-event/8000
      a=fmtp:100 0-16
      a=silenceSupp:off - - - -
      a=ptime:20
      a=sendrecv
    ]]>
</send>


<recv response="100" optional="true">
</recv>

<recv response="180" optional="true">
</recv>




<recv response="200" rtd="true" crlf="true">
</recv>



<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=[call_number]
      To: sut <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>






<pause milliseconds="5000"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>

<pause milliseconds="1000"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>
<pause milliseconds="1000"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>
<pause milliseconds="500"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>
<pause milliseconds="500"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>
<pause milliseconds="500"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>
<pause milliseconds="500"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>
<pause milliseconds="500"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>
<pause milliseconds="500"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>
<pause milliseconds="500"/>


<nop>
    <action>
      <exec play_pcap_audio="dtmf_2833_1.pcap"/>
    </action>
</nop>

<pause milliseconds="2000"/>


<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=[call_number]
      To: sut <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>


<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>


<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>

Cheers,

Michael

On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

That's a pretty small problem description to be so sure about something.
It would probably be better to capture some evidence of the exact problem you are having since we are using computers and we need to see the computers in action doing something specifically incorrect to diagnose any sort of problem. Take the time to describe the origin and destination of your calls, the call flow, the hardware in use on both ends of the call, detailed console logs on debug level, (maybe even uncomment the 2833 debug ifded in switch_rtp.c) and gather something to go on besides "I seem to be losing dtmf) maybe a packect capture of the networking interface on both ends of these calls.

Also problems should be reported to http://jira.freeswitch.org not this mailing list.
Save us a step if you report a jira and provide all the info above or we will just have to ask for it again.

On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop <michaelt@voxcore.voxtelecom.co.za> wrote:

Hi All,

I have an issue that when my call volumes on my FS IVR box > 30 calls DTMF digits are lost (using RFC2833). It is definitely load related as it all works perfectly under 30 calls.

Any pointers or a solution to the problem?

Thanks,

Michael
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org